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2011-02-23 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.18-rc1 Released.
2011-02-22 15:37 +0000 [r308528] Andrew Latham <lathama@gmail.com>
* main/http.c: Add HTTP URI log, use ast_debug for console logging
Guessed the log levels based on info that level 3 is the soft
roof. Can we create a page / document to define the levels?
2011-02-21 15:00 +0000 [r308414] Matthew Nicholson <mnicholson@digium.com>
* main/udptl.c, /: Merged revisions 308413 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r308413 | mnicholson | 2011-02-21 08:57:15 -0600 (Mon, 21 Feb
2011) | 5 lines Properly check the bounds of arrays when decoding
UDPTL packets. Also, remove broken support for receiving UDPTL
packets larger than 16k. That shouldn't ever happen anyway.
AST-2011-002 FAX-281 ........
2011-02-19 14:03 +0000 [r308329] Andrew Latham <lathama@gmail.com>
* main/http.c: Add CSS MIME Type Modern browsers are checking for
the MIME Type of pages and in some cases will not load a file if
the type is wrong.
2011-02-15 23:33 +0000 [r308007] Jason Parker <jparker@digium.com>
* apps/app_queue.c, /: Merged revisions 308002 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r308002 | qwell | 2011-02-15 17:32:20 -0600 (Tue, 15 Feb 2011) |
10 lines Fix regression that changed behavior of queues when
ringing a queue member. This reverts r298596, which was to fix a
highly bizarre and contrived issue with a queue member that
called into his own queue being transferred back into his own
queue. I couldn't reproduce that issue in any way. I think one of
the other recent transfer fixes actually fixed this. (closes
issue #18747) Reported by: vrban ........
2011-02-15 07:01 +0000 [r307792-307836] Tilghman Lesher <tilghman@meg.abyt.es>
* funcs/func_odbc.c: Need to retrieve the rows affected before
using the associated variable. (closes issue #18795) Reported by:
irroot Patches: 20110211__issue18795.diff.txt uploaded by
tilghman (license 14) Tested by: tilghman
* res/res_odbc.c: Increment usage count at first reference, to
avoid a race condition with many threads creating connections all
at once. (issue #18156) Reported by: asgaroth Patches:
20110214__issue18156.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
2011-02-11 01:02 +0000 [r307624] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 307623 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r307623 | rmudgett | 2011-02-10 18:29:17 -0600 (Thu, 10
Feb 2011) | 13 lines Reentrancy problem if outgoing call gets
different B channel than requested. The chan_dahdi
pri_fixup_principle() routine needs to protect the Asterisk
channel with the channel lock when it changes the technology
private pointer to a new private structure. * Added lock
protection while pri_fixup_principle() moves a call from one
private structure to another. * Made some pri_fixup_principle()
messages more meaningful. Partial backport from v1.8 -r300714.
........
2011-02-10 22:35 +0000 [r307535] Jason Parker <jparker@digium.com>
* main/asterisk.c, contrib/init.d/rc.debian.asterisk, /: Merged
revisions 307534 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r307534 | qwell | 2011-02-10 16:33:09 -0600 (Thu, 10 Feb 2011) |
8 lines Remove color when executing commands via a remote
console. Essentially this makes '-x' imply '-n' on rasterisk.
This was done in a different and incomplete way previously, which
I'm reverting here. (issue #18776) Reported by: alecdavis
........
2011-02-09 21:48 +0000 [r307316] Andrew Latham <lathama@gmail.com>
* contrib/init.d/rc.debian.asterisk: Disable color during running
test (closes issue #18776) Reported by: alecdavis Patches:
ast_deb_init.diff uploaded by lathama (license 1028) Tested by:
andrel, lathama
2011-02-09 19:52 +0000 [r307227] Jeff Peeler <jpeeler@digium.com>
* main/features.c: Make sure to set parking dial context for
non-default parking lots. Since parking_con_dial isn't settable,
set all parking lots to "park-dial". (closes issue #17946)
Reported by: bluecrow76 Patches:
asterisk-1.8.0-beta4-multipark-fixes-2010SEP02.diff uploaded by
bluecrow76 (license 270) modified by me
2011-02-08 20:14 +0000 [r306973] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c: Merged revisions 306972 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r306972 | twilson | 2011-02-08 12:05:13 -0800 (Tue, 08 Feb 2011)
| 2 lines Fix comparison for REFER Replaces tags with
pedantic=yes ........
2011-02-08 19:41 +0000 [r306865-306966] Jeff Peeler <jpeeler@digium.com>
* apps/app_voicemail.c, /: Merged revisions 306965 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r306965 | jpeeler | 2011-02-08 13:40:58 -0600 (Tue, 08
Feb 2011) | 1 line fix this line again ........
* apps/app_voicemail.c, /: Merged revisions 306960 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r306960 | jpeeler | 2011-02-08 13:18:50 -0600 (Tue, 08
Feb 2011) | 9 lines Backup file storing message duration is not
used with IMAP_STORAGE, remove code. The message duration is
stored in the body of the email when using IMAP_STORAGE, so
nothing needs to happen with the backup file. (closes issue
#18718) Reported by: kerframil ........
* apps/app_voicemail.c, /: Merged revisions 306864 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r306864 | jpeeler | 2011-02-08 10:19:17 -0600 (Tue, 08
Feb 2011) | 1 line make this safer and fully correct, pointed out
by Steve Davis ........
2011-02-07 22:40 +0000 [r306618-306673] Terry Wilson <twilson@digium.com>
* /, main/features.c: Merged revisions 306672 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r306672 | twilson | 2011-02-07 14:35:20 -0800 (Mon, 07 Feb 2011)
| 10 lines Don't try to pickup a call in the middle of a
masquerade If A calls B which doesn't answer and C & D both try
to do a call pickup, it is possible for ast_pickup_call to answer
the call, then fail to masquerade one of the calls because the
other one is already in the process of masquerading. This patch
checks to see if the channel is in the process of masquerading
before call before selecting it for a pickup. Review:
https://reviewboard.asterisk.org/r/1094/ ........
* /, channels/chan_sip.c: Merged revisions 306617 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r306617 | twilson | 2011-02-07 13:51:43 -0800 (Mon, 07 Feb 2011)
| 10 lines Don't allow a REFER w/replaces to replace its own
dialog Asterisk currently accepts a REFER with a Refer-To with an
embedded Replaces header that matches the dialog of the REFER.
This would be a situation like A calls B, A calls C, A transfers
B to A, which is just silly. This patch makes the transfer fail
instead of making Asterisk freak out and forget to hang other
channels up. Review: https://reviewboard.asterisk.org/r/1093/
........
2011-02-04 19:21 +0000 [r306346] Jason Parker <jparker@digium.com>
* apps/app_queue.c: Don't fallthrough to 'unknown' in the 'ringing'
case. This could cause improper exits from the queue. (closes
issue #18499) Reported by: zaltar Patches: app_queue.patch
uploaded by zaltar (license 1148)
2011-02-03 20:56 +0000 [r306126] Terry Wilson <twilson@digium.com>
* channels/chan_local.c, /: Merged revisions 306119 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r306119 | twilson | 2011-02-03 12:36:34 -0800 (Thu, 03
Feb 2011) | 9 lines Set hangup cause in local_hangup When a call
involves a local channel (like SIP -> Local -> SIP), the hangup
cause was not being set. This resulted in SIP channels sometimes
getting a 503 error instead of a 486 when the far side sent a
busy. In Asterisk 1.8+ this also can cause issues with CCSS that
involve a local channel. This patch sets the hangupcause for one
side of the local channel to the other in local_hangup for
outbound calls. ........
2011-02-03 20:49 +0000 [r306123] Jeff Peeler <jpeeler@digium.com>
* main/features.c: Set exception on channel in parking thread when
POLLPRI event detected. This is done just to make the code be
equivalent to the old select code. As noted in 303106 the same
issue was already fixed in this branch, but the exception was not
set on the channel in the case of POLLPRI. The reason that this
did not cause a problem here is because in 122923 the check in
__ast_read to check the exception flag was removed. (related to
#18637)
2011-02-03 15:41 +0000 [r305985] Andrew Latham <lathama@gmail.com>
* phoneprov/snom-mac.xml (added), configs/phoneprov.conf.sample:
res_phoneprov add snom 300, 320, 360, 370, 820, 821, 870 support
(issue #18713) Reported by: lathama Patches: snom_dir.diff
uploaded by lathama (license 1028) Tested by: lathama
2011-02-03 00:15 +0000 [r305889] Richard Mudgett <rmudgett@digium.com>
* main/channel.c, main/manager.c, /, channels/chan_sip.c,
apps/app_sendtext.c: Merged revisions 305888 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r305888 | rmudgett | 2011-02-02 18:02:43 -0600 (Wed, 02 Feb 2011)
| 8 lines Minor AST_FRAME_TEXT related issues. * Include the null
terminator in the buffer length. When the frame is queued it is
copied. If the null terminator is not part of the frame buffer
length, the receiver could see garbage appended onto it. * Add
channel lock protection with ast_sendtext(). * Fixed AMI SendText
action ast_sendtext() return value check. ........
2011-02-02 14:40 +0000 [r305648-305752] Andrew Latham <lathama@gmail.com>
* channels/chan_sip.c: Replace link to old doc with new wiki page.
Link to
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
* configs/sip.conf.sample: SIP Configuration Documentation sip show
settings reports qualifyfreq in milliseconds. sip.conf configures
qualifyfreg in seconds.
2011-02-01 17:02 +0000 [r305472] Jason Parker <jparker@digium.com>
* res/res_musiconhold.c, /: Merged revisions 305471 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r305471 | qwell | 2011-02-01 11:00:55 -0600 (Tue, 01 Feb
2011) | 9 lines Close file descriptor for timing source when a
MOH class gets destroyed. (closes issue #18457) Reported by:
mcallist Patches: 18457-closetimer.diff uploaded by qwell
(license 4) 18457-closetimer_trunk.diff uploaded by qwell
(license 4) Tested by: qwell, loloski ........
2011-01-31 23:50 +0000 [r305342] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 305341 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r305341 | rmudgett | 2011-01-31 17:45:58 -0600 (Mon, 31
Jan 2011) | 7 lines Obtain the pri lock for PRI queue counters.
Need to obtain the pri lock when calling pri_dump_info_str() to
avoid a reentrancy problem when calculating the Q.921 Q count
statistic. JIRA AST-484 ........
2011-01-31 22:59 +0000 [r305130-305253] Jason Parker <jparker@digium.com>
* apps/app_dial.c, /, channels/chan_sip.c: Merged revisions 305252
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r305252 | qwell | 2011-01-31 16:56:54 -0600 (Mon, 31 Jan 2011) |
10 lines Prevent a crash when dialing a technology with no
destination (ex: Dial(SIP/)) chan_iax2 and other channel drivers
already had code to prevent this. The attempt that app_dial was
making to prevent it was not correct, so I fixed that. (closes
issue #18371) Reported by: gbour Patches: 18371.patch uploaded by
gbour (license 1162) ........
* res/res_musiconhold.c, /: Merged revisions 305129 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r305129 | qwell | 2011-01-31 14:56:25 -0600 (Mon, 31 Jan
2011) | 2 lines Set file descriptors to -1 on creation, so that
we don't see weirdness later. ........
2011-01-31 13:52 +0000 [r305082] Andrew Latham <lathama@gmail.com>
* main/http.c: Asterisk HTTP response Content-type Address content
type for BSD and other platforms (closes issue #18456) Reported
by: alexo Patches: asterisk18_http.patch uploaded by alexo
(license 1175) Tested by: alexo
2011-01-31 07:25 +0000 [r304978] Tilghman Lesher <tilghman@meg.abyt.es>
* apps/app_voicemail.c, /: Merged revisions 304952 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r304952 | tilghman | 2011-01-31 00:54:45 -0600 (Mon, 31
Jan 2011) | 2 lines Fix compilation when ODBC_STORAGE is defined.
........
2011-01-29 23:05 +0000 [r304659-304865] Sean Bright <sean@malleable.com>
* res/res_config_ldap.c: Plug some memory leaks in the LDAP
realtime driver. (closes issue #18435) Reported by: zaltar
Patches: res_config_ldap.patch uploaded by zaltar (license 1148)
* apps/app_meetme.c: If we fail to allocate our announcement
objects, make sure we don't leak objects. The majority of this
patch was committed already in r304726 and r304729. (issue
#18225) Reported by: kenji (issue #18444) Reported by: junky
(closes issue #18343) Reported by: kobaz Patches:
meetme-refs.diff uploaded by kobaz (license 834)
* apps/app_meetme.c: When we pass the S() or L() options to MeetMe,
make sure that we honor C as well. Without this patch, if the
user was kicked from the conference via the S() or L() mechanism,
we would just hang up on them even if we also passed C (continue
in dialplan when kicked). With this patch we honor the C flag in
those cases. (closes issue #17317) Reported by: var
* apps/app_meetme.c: Make sure that we unref the correct object
when ejecting the most recent caller. Currently, when we kick the
last user to enter, we decrement our own reference count which
results in a crash when we kick another user or when we exit the
conference ourselves. This will fix #18225 in 1.8 and trunk, but
that particular bug does not exist in 1.6.2. (closes issue
#18225) Reported by: kenji Patches: issue18225.patch uploaded by
seanbright (license 71) Tested by: seanbright
* apps/app_meetme.c: Fix user reference leak in MeetMe. We were
unlinking the user from the conferences user container, but not
decrementing the reference count of the user as well, resulting
in a leak. (closes issue #18444) Reported by: junky Tested by:
seanbright
* apps/app_meetme.c: Revert part of the previous commit that snuck
in.
* apps/app_meetme.c: Don't leak references if we can't create a
pseudo channel for mixing in MeetMe. If there was a problem
allocating a pseudo channel when building our meetme, we weren't
destroying our user container or destroying the mutexes that we
created.
2011-01-27 17:01 +0000 [r304461-304465] Jason Parker <jparker@digium.com>
* /, configure, configure.ac: Merged revisions 304464 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r304464 | qwell | 2011-01-27 10:57:46 -0600 (Thu, 27 Jan
2011) | 9 lines Fix default prefix=/usr regression on non-Linux
systems. This partially reverts a change made in branches/1.4/
r267759, which will cause issue #17013 to be reopened. This issue
was pointed out by a user on #asterisk, who helpfully discovered
that paths were being set incorrectly. To truly understand what
was wrong, one should run: svn diff --force -c<this revision>
configure ........
* /, configure: Merged revisions 304460 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r304460 | qwell | 2011-01-27 10:47:03 -0600 (Thu, 27 Jan 2011) |
1 line Rerun bootstrap.sh with no changes, so that it is more
obvious what my next commit changes. ........
2011-01-26 22:26 +0000 [r304338] Jeff Peeler <jpeeler@digium.com>
* main/features.c: Change delimiter used internally for
GOTO_ON_BLINDXFR to commas to match 76703.
2011-01-26 21:02 +0000 [r304250] Mark Michelson <mmichelson@digium.com>
* main/udptl.c, /: Merged revisions 304242 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r304242 | mmichelson | 2011-01-26 14:38:37 -0600 (Wed, 26 Jan
2011) | 3 lines Get rid of unused 'verbose' field in ast_udptl
........
2011-01-26 21:01 +0000 [r304244-304249] Matthew Nicholson <mnicholson@digium.com>
* /: Merged revisions 304247 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r304247 | mnicholson | 2011-01-26 15:00:15 -0600 (Wed, 26 Jan
2011) | 2 lines Convert from network to host byte ordering before
checking if an IP is a multicast address. ........
* /, channels/chan_sip.c: Merged revisions 304241 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r304241 | mnicholson | 2011-01-26 14:38:22 -0600 (Wed, 26 Jan
2011) | 6 lines This patch modifies chan_sip to route responses
to the address the request came from. It also modifies chan_sip
to respect the maddr parameter in the Via header. ABE-2664
Review: https://reviewboard.asterisk.org/r/1059/ ........
2011-01-26 20:22 +0000 [r304181] Sean Bright <sean@malleable.com>
* /, configs/queues.conf.sample: Merged revisions 304159 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r304159 | seanbright | 2011-01-26 15:18:29 -0500 (Wed, 26 Jan
2011) | 1 line Make sure the sample queues.conf is properly
commented. ........
2011-01-26 19:38 +0000 [r304149] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: Merged revisions 304148 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
.......... r304148 | rmudgett | 2011-01-26 13:23:46 -0600 (Wed,
26 Jan 2011) | 2 lines Update documentation for
DAHDISendCallreroutingFacility() application. ..........
2011-01-26 01:24 +0000 [r304096] Sean Bright <sean@malleable.com>
* main/file.c: Per the man page, setvbuf() must be called before
any other operation on an open file. We use setvbuf() to
associate a buffer with a stream, but we have already written to
the open file. This works (by chance) on Linux, but fails on
other platforms, such as OpenSolaris. (closes issue #16610)
Reported by: bklang Patches: setvbuf.patch uploaded by crjw
(license 963) Tested by: bklang, asgaroth, efutch
2011-01-25 23:25 +0000 [r304006] Richard Mudgett <rmudgett@digium.com>
* /, main/features.c: Merged revisions 304005 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r304005 | rmudgett | 2011-01-25 17:21:09 -0600 (Tue, 25 Jan 2011)
| 8 lines DTMF attended transfers sometimes fail for no apparent
reason. The loop in feature_request_and_dial() can exit when
Party C has answered without processing an AST_CONTROL_ANSWER.
Also sometimes an AST_CONTROL_ANSWER never happens even though
Party C has answered. Don't hangup Party C if he is up or we
receive an AST_CONTROL_ANSWER. ........
2011-01-25 22:02 +0000 [r303960] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c: Merged revisions 303906 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r303906 | twilson | 2011-01-25 14:50:59 -0600 (Tue, 25 Jan 2011)
| 16 lines Guard against retransmitting BYEs indefinitely In the
case of an attended transfer (A calls B, A atxfers to C) where A
becomes unreachable before replying to Asterisk's BYE, Asterisk
can sometimes retransmit the BYE indefinitely. This is because
__sip_autodestruct tests p->refer && !ast_test_flag(&p->flags[0],
SIP_ALREADYGONE and will then transmit a BYE. When this BYE times
out, it will not ever be marked as ALREADYGONE, so when
__sip_autodestruct is called again, we end up starting the cycle
over. This patch adds a call to sip_alreadygone(pkt->owner) in
retrans_pkt in the case of a BYE that has timed out. This should
prevent Asterisk from trying to transmit new BYE messages in the
future. Review: https://reviewboard.asterisk.org/r/1077/ ........
2011-01-25 18:41 +0000 [r303858] Tilghman Lesher <tilghman@meg.abyt.es>
* channels/chan_sip.c: Fix "sip show user <tab>", so that it
actually shows results, instead of just completing the last
entry. (closes issue #16675) Reported by: pj
2011-01-25 17:42 +0000 [r303769] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 303765 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r303765 | rmudgett | 2011-01-25 11:36:50 -0600 (Tue, 25
Jan 2011) | 40 lines Sending out unnecessary PROCEEDING messages
breaks overlap dialing. Issue #16789 was a good idea.
Unfortunately, it breaks overlap dialing through Asterisk. There
is not enough information available at this point to know if
dialing is complete. The ast_exists_extension(),
ast_matchmore_extension(), and ast_canmatch_extension() calls are
not adequate to detect a dial through extension pattern of "_9!".
Workaround is to use the dialplan Proceeding() application early
in non-dial through extensions. * Effectively revert issue
#16789. * Allow outgoing overlap dialing to hear dialtone and
other early media. A PROGRESS "inband-information is now
available" message is now sent after the SETUP_ACKNOWLEDGE
message for non-digital calls. An AST_CONTROL_PROGRESS is now
generated for incoming SETUP_ACKNOWLEDGE messages for non-digital
calls. * Handling of the AST_CONTROL_CONGESTION in
chan_dahdi/sig_pri was inconsistent with the cause codes. * Added
better protection from sending out of sequence messages by
combining several flags into a single enum value representing
call progress level. * Added diagnostic messages for deferred
overlap digits handling corner cases. (closes issue #17085)
Reported by: shawkris (closes issue #18509) Reported by: wimpy
Patches: issue18509_early_media_v1.8_v3.patch uploaded by
rmudgett (license 664) Expanded upon
issue18509_early_media_v1.8_v3.patch to include analog and SS7
because of backporting requirements. Tested by: wimpy, rmudgett
........
2011-01-25 16:59 +0000 [r303677] Jeff Peeler <jpeeler@digium.com>
* apps/app_voicemail.c, /: Merged revisions 303676 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r303676 | jpeeler | 2011-01-25 10:58:29 -0600 (Tue, 25
Jan 2011) | 20 lines Fix voicemail sequencing for file based
storage. A previous change was made to account for when the
number of voicemail messages exceeds the max limit to be handled
properly, but it caused gaps in the messages to not be properly
handled. This has now been resolved. In later non 1.4 branches,
it appears that resequencing wasn't even occurring due from what
appears and accidental code removal. (closes issue #18498)
Reported by: JJCinAZ Patches: bug18498v2.patch uploaded by
jpeeler (license 325) (closes issue #18486) Reported by: bluefox
Patches: bug18486.patch uploaded by jpeeler (license 325)
........
2011-01-24 20:49 +0000 [r303548] Russell Bryant <russell@digium.com>
* main/channel.c, main/pbx.c, /, apps/app_meetme.c,
main/features.c, include/asterisk/channel.h: Merged revisions
303546 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r303546 | russell | 2011-01-24 14:32:21 -0600 (Mon, 24 Jan 2011)
| 31 lines Fix channel redirect out of MeetMe() and other issues
with channel softhangup. Mantis issue #18585 reports that a
channel redirect out of MeetMe() stopped working properly. This
issue includes a patch that resolves the issue by removing a call
to ast_check_hangup() from app_meetme.c. I left that in my patch,
as it doesn't need to be there. However, the rest of the patch
fixes this problem with or without the change to app_meetme. The
key difference between what happens before and after this patch
is the effect of the END_OF_Q control frame. After END_OF_Q is
hit in ast_read(), ast_read() will return NULL. With the
ast_check_hangup() removed, app_meetme sees this which causes it
to exit as intended. Checking ast_check_hangup() caused
app_meetme to exit earlier in the process, and the target of the
redirect saw the condition where ast_read() returned NULL.
Removing ast_check_hangup() works around the issue in app_meetme,
but doesn't solve the issue if another application did the same
thing. There are also other edge cases where if an application
finishes at the same time that a redirect happens, the target of
the redirect will think that the channel hung up. So, I made some
changes in pbx.c to resolve it at a deeper level. There are
already places that unset the SOFTHANGUP_ASYNCGOTO flag in an
attempt to abort the hangup process. My patch extends this to
remove the END_OF_Q frame from the channel's read queue, making
the "abort hangup" more complete. This same technique was used in
every place where a softhangup flag was cleared. (closes issue
#18585) Reported by: oej Tested by: oej, wedhorn, russell Review:
https://reviewboard.asterisk.org/r/1082/ ........
2011-01-21 21:48 +0000 [r303285] Jason Parker <jparker@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 303284 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r303284 | qwell | 2011-01-21 15:45:34 -0600 (Fri, 21 Jan
2011) | 8 lines Reset configuration before parsing users.conf.
Some values configured in chan_dahdi.conf were able to leak in to
users.conf configuration. This was surprising users, and
potentially setting non-sane "defaults". ASTNOW-125 ........
2011-01-21 16:12 +0000 [r303273] Leif Madsen <lmadsen@digium.com>
* apps/app_dial.c: Fix changes to L() flag in Dial(). Tony
Mountifield pointed out an error I had in my patch. I was a bit
too aggressive on changing 'seconds' to 'milliseconds'. So I
decided to do some additioanl testing and have no changed just
the appropriate lines. One line says milliseconds, and the other
says seconds. Probably should change this to be either just
seconds or milliseconds, but I've spent too much time on this
already :) (issue #18264)
2011-01-20 19:56 +0000 [r303106] Shaun Ruffell <sruffell@digium.com>
* main/features.c: main/features: Use POLLPRI when waiting for
events on parked channels. This change resolves a regression in
the 1.6.2 when converting from select to poll. The DAHDI timers
use POLLPRI to indicate that the timer fired, but features was
not waiting for that flag. The result was no audio for MOH when a
call was parked and res_timing_dahdi was in use. This patch is
slightly modified from the one on the mantis issue. It does not
set an exception on the channel if the POLLPRI flag is set.
(closes issue #18262) Reported by: francesco_r Patches:
patch_park_moh-trunk-2.txt uploaded by cjacobsen (license 1029)
Tested by: francesco_r, rfrantik, one47
2011-01-20 17:07 +0000 [r303008] Jeff Peeler <jpeeler@digium.com>
* apps/app_queue.c, /, configs/queues.conf.sample: Merged revisions
303007 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r303007 | jpeeler | 2011-01-20 11:04:08 -0600 (Thu, 20 Jan 2011)
| 8 lines Add new queue strategy to preserve behavior for when
queue members moved to ao2. Add queue strategy called "rrordered"
to mimic old behavior from when queue members were stored in a
linked list. ABE-2707 ........
2011-01-20 16:11 +0000 [r302920] Russell Bryant <russell@digium.com>
* apps/app_privacy.c: Resolve a compiler warning.
2011-01-20 15:42 +0000 [r302917] Leif Madsen <lmadsen@digium.com>
* apps/app_dial.c, /: Option L() is milliseconds, not seconds. >
Change the verbose output of option L() to say milliseconds and
not seconds > as the value is in milliseconds. > > (closes issue
#18264) > Reported by: jacco > Patches: > app_dial_patch.txt
uploaded by lmadsen (license 10)
2011-01-19 23:47 +0000 [r302833] Sean Bright <sean@malleable.com>
* apps/app_voicemail.c: Support greetingsfolder as documented in
voicemail.conf.sample. (closes issue #17870) Reported by:
edhorton Patches:
__20100816-app_voicemail-greetingsfolder-support.txt uploaded by
lmadsen (license 10)
2011-01-19 23:06 +0000 [r302788] Russell Bryant <russell@digium.com>
* main/manager.c: Turn a noisy verbose message into a debug
message. This can drown your console if you're using the AMI over
HTTP.
2011-01-19 21:25 +0000 [r302693] Richard Mudgett <rmudgett@digium.com>
* /, main/features.c: Merged revisions 302671 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r302671 | rmudgett | 2011-01-19 15:21:56 -0600 (Wed, 19 Jan 2011)
| 15 lines DTMF transfer plays the wrong sounds for wrong number
or other call failure. * Set the default for features.conf.sample
xferfailsound option to "beeperr" as documented instead of
"pbx-invalid" and corrected the use of it in DTMF blind transfer
(#1). * Improved DTMF blind transfer handling of wrong numbers.
Most of the concerns in this issue were taken care of by the
patch for issue 17999: Issues with DTMF triggered attended
transfers. (closes issue #18379) Reported by: gincantalupo Tested
by: rmudgett ........
2011-01-19 21:22 +0000 [r302599-302675] Tilghman Lesher <tilghman@meg.abyt.es>
* include/asterisk/astdb.h, /: Merged revisions 302663 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r302663 | tilghman | 2011-01-19 15:20:28 -0600 (Wed, 19
Jan 2011) | 2 lines Add some API documentation ........
* main/app.c: Kill zombies. When we ast_safe_fork() with a non-zero
argument, we're expected to reap our own zombies. On a zero
argument, however, the zombies are only reaped when there aren't
any non-zero forked children alive. At other times, we accumulate
zombies. This code is forward ported from res_agi in 1.4, so that
forked children are always reaped, thus preventing an
accumulation of zombie processes. (closes issue #18515) Reported
by: ernied Patches: 20101221__issue18515.diff.txt uploaded by
tilghman (license 14) Tested by: ernied
2011-01-19 19:02 +0000 [r302504-302554] Sean Bright <sean@malleable.com>
* main/utils.c: Don't call strlen() when we only need to look at
the next character or two. (closes issue #18042) Reported by:
wdoekes Patches: astsvn-inefficient-ast-uri-decode.patch uploaded
by wdoekes (license 717)
* main/features.c: Remove an extraneous \r\n at the end of a
parking manager events. (closes issue #18363) Reported by:
clegall_proformatique Patches:
asterisk_1.8_295998_parking_manager_events_format.patch uploaded
by clegall proformatique (license 1139)
* res/res_agi.c: Properly handle partial reads from fgets() when
handling AGIs. When fgets() failed with EAGAIN, we were
continually decrementing the available space left in our buffer,
resulting in botched command handling. (closes issue #16032)
Reported by: notahat Patches: agi_buffer_patch2.diff uploaded by
fnordian (license 110)
* main/utils.c: Make sure that h_length is set when we
short-circuit out of ast_gethostbyname. (closes issue #16135)
Reported by: thedavidfactor Patches: utils.patch uploaded by
thedavidfactor (license 903)
2011-01-19 17:08 +0000 [r302461] Paul Belanger <pabelanger@digium.com>
* res/res_timing_timerfd.c: Handle 'Resource temporarily
unavailable' error more gracefully.
2011-01-19 15:52 +0000 [r302416] Sean Bright <sean@malleable.com>
* configs/extensions.conf.sample: Remove references to
priorityjumping from the sample extensions.conf. Priority jumping
was removed from pbx_config in r68970. (closes issue #18622)
Reported by: kshumard Patches: extensions.conf.sample.patch
uploaded by kshumard (license 92)
2011-01-18 21:40 +0000 [r302313] Matthew Nicholson <mnicholson@digium.com>
* /, channels/chan_sip.c: Merged revisions 302311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r302311 | mnicholson | 2011-01-18 15:35:03 -0600 (Tue, 18 Jan
2011) | 4 lines URI encode the user part of the contact header.
ABE-2705 ........
2011-01-18 20:13 +0000 [r302265] Jeff Peeler <jpeeler@digium.com>
* main/pbx.c: Convert device state callbacks to ao2 objects to fix
a deadlock in chan_sip. Lock scenario presented here: Thread 1
holds ast_rdlock_contexts &conlock holds handle_statechange hints
holds handle_statechange hint waiting for cb_extensionstate
Locked Here: chan_sip.c line 7428 (find_call) Thread 2 holds
handle_request_do &netlock holds find_call sip_pvt_ptr waiting
for ast_rdlock_contexts &conlock Locked Here: pbx.c line 9911
(ast_rdlock_contexts) Chan_sip has an established locking order
of locking the sip_pvt and then getting the context lock. So the
as stated by the summary, the operations in thread 2 have been
modified to no longer require the context lock. (closes issue
#18310) Reported by: one47 Patches: statecbs_ao2.mk2.patch
uploaded by one47 (license 23), modified by me Review:
https://reviewboard.asterisk.org/r/1072/
2011-01-18 18:07 +0000 [r302173] Richard Mudgett <rmudgett@digium.com>
* /, main/features.c: Merged revisions 302172 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r302172 | rmudgett | 2011-01-18 12:04:36 -0600 (Tue, 18 Jan 2011)
| 88 lines Issues with DTMF triggered attended transfers. Issue
#17999 1) A calls B. B answers. 2) B using DTMF dial *2 (code in
features.conf for attended transfer). 3) A hears MOH. B dial
number C 4) C ringing. A hears MOH. 5) B hangup. A still hears
MOH. C ringing. 6) A hangup. C still ringing until
"atxfernoanswertimeout" expires. For v1.4 C will ring forever
until C answers the dead line. (Issue #17096) Problem: When A and
B hangup, C is still ringing. Issue #18395 SIP call limit of B is
1 1. A call B, B answered 2. B *2(atxfer) call C 3. B hangup, C
ringing 4. Timeout waiting for C to answer 5. Recall to B fails
because B has reached its call limit. Because B reached its call
limit, it cannot do anything until the transfer it started
completes. Issue #17273 Same scenario as issue 18395 but party B
is an FXS port. Party B cannot do anything until the transfer it
started completes. If B goes back off hook before C answers, B
hears ringback instead of the expected dialtone. ********** Note
for the issue #17273 and #18395 fix: DTMF attended transfer works
within the channel bridge. Unfortunately, when either party A or
B in the channel bridge hangs up, that channel is not completely
hung up until the transfer completes. This is a real problem
depending upon the channel technology involved. For chan_dahdi,
the channel is crippled until the hangup is complete. Either the
channel is not useable (analog) or the protocol disconnect
messages are held up (PRI/BRI/SS7) and the media is not released.
For chan_sip, a call limit of one is going to block that endpoint
from any further calls until the hangup is complete. For party A
this is a minor problem. The party A channel will only be in this
condition while party B is dialing and when party B and C are
conferring. The conversation between party B and C is expected to
be a short one. Party B is either asking a question of party C or
announcing party A. Also party A does not have much incentive to
hangup at this point. For party B this can be a major problem
during a blonde transfer. (A blonde transfer is our term for an
attended transfer that is converted into a blind transfer. :))
Party B could be the operator. When party B hangs up, he assumes
that he is out of the original call entirely. The party B channel
will be in this condition while party C is ringing, while
attempting to recall party B, and while waiting between call
attempts. WARNING: The ATXFER_NULL_TECH conditional is a hack to
fix the problem. It will replace the party B channel technology
with a NULL channel driver to complete hanging up the party B
channel technology. The consequences of this code is that the 'h'
extension will not be able to access any channel technology
specific information like SIP statistics for the call.
ATXFER_NULL_TECH is not defined by default. ********** (closes
issue #17999) Reported by: iskatel Tested by: rmudgett JIRA
SWP-2246 (closes issue #17096) Reported by: gelo Tested by:
rmudgett JIRA SWP-1192 (closes issue #18395) Reported by:
shihchuan Tested by: rmudgett (closes issue #17273) Reported by:
grecco Tested by: rmudgett Review:
https://reviewboard.asterisk.org/r/1047/ ........
2011-01-17 16:53 +0000 [r302049] Terry Wilson <twilson@digium.com>
* channels/chan_sip.c: Merged revisions 293493 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 [^] ........
r293493 | twilson | 2010-11-01 09:58:00 -0500 (Mon, 01 Nov 2010)
| 14 lines Only offer codecs both sides support for directmedia
When using directmedia, Asterisk needs to limit the codecs
offered to just the ones that both sides recognize, otherwise
they may end up sending audio that the other side doesn't
understand. (closes issue 0017403) Reported by: one47 Patches:
sip_codecs_simplified4 uploaded by one47 (license 23) Tested by:
one47, falves11 Review: https://reviewboard.asterisk.org/r/967/
[^] ........ Backporting a bugfix that should have been included.
2011-02-22 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.17 Released.
* Merged in changes related to AST-2011-002
2011-02-16 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.17-rc3 Released.
------------------------------------------------------------------------
r308002 | qwell | 2011-02-15 17:32:21 -0600 (Tue, 15 Feb 2011) | 10
lines
Fix regression that changed behavior of queues when ringing a queue
member.
This reverts r298596, which was to fix a highly bizarre and contrived
issue with a queue member that called into his own queue being
transferred back into his own queue. I couldn't reproduce that issue
in any way. I think one of the other recent transfer fixes actually
fixed this.
(closes issue 0018747)
Reported by: vrban
------------------------------------------------------------------------
2011-01-20 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.17-rc2 Released.
------------------------------------------------------------------------
r302172 | rmudgett | 2011-01-18 12:04:37 -0600 (Tue, 18 Jan 2011) | 88
lines
Issues with DTMF triggered attended transfers.
Issue 0017999
1) A calls B. B answers.
2) B using DTMF dial *2 (code in features.conf for attended transfer).
3) A hears MOH. B dial number C
4) C ringing. A hears MOH.
5) B hangup. A still hears MOH. C ringing.
6) A hangup. C still ringing until "atxfernoanswertimeout" expires.
For v1.4 C will ring forever until C answers the dead line. (Issue
0017096)
Problem: When A and B hangup, C is still ringing.
Issue 0018395
SIP call limit of B is 1
1. A call B, B answered
2. B *2(atxfer) call C
3. B hangup, C ringing
4. Timeout waiting for C to answer
5. Recall to B fails because B has reached its call limit.
Because B reached its call limit, it cannot do anything until the
transfer
it started completes.
Issue 0017273
Same scenario as issue 18395 but party B is an FXS port. Party B
cannot
do anything until the transfer it started completes. If B goes back
off
hook before C answers, B hears ringback instead of the expected
dialtone.
**********
Note for the issue 0017273 and 0018395 fix:
DTMF attended transfer works within the channel bridge. Unfortunately,
when either party A or B in the channel bridge hangs up, that channel
is
not completely hung up until the transfer completes. This is a real
problem depending upon the channel technology involved.
For chan_dahdi, the channel is crippled until the hangup is complete.
Either the channel is not useable (analog) or the protocol disconnect
messages are held up (PRI/BRI/SS7) and the media is not released.
For chan_sip, a call limit of one is going to block that endpoint from
any
further calls until the hangup is complete.
For party A this is a minor problem. The party A channel will only be
in
this condition while party B is dialing and when party B and C are
conferring. The conversation between party B and C is expected to be a
short one. Party B is either asking a question of party C or
announcing
party A. Also party A does not have much incentive to hangup at this
point.
For party B this can be a major problem during a blonde transfer. (A
blonde transfer is our term for an attended transfer that is converted
into a blind transfer. :)) Party B could be the operator. When party B
hangs up, he assumes that he is out of the original call entirely. The
party B channel will be in this condition while party C is ringing,
while
attempting to recall party B, and while waiting between call attempts.
WARNING:
The ATXFER_NULL_TECH conditional is a hack to fix the problem. It will
replace the party B channel technology with a NULL channel driver to
complete hanging up the party B channel technology. The consequences
of
this code is that the 'h' extension will not be able to access any
channel
technology specific information like SIP statistics for the call.
ATXFER_NULL_TECH is not defined by default.
**********
(closes issue 0017999)
Reported by: iskatel
Tested by: rmudgett
JIRA SWP-2246
(closes issue 0017096)
Reported by: gelo
Tested by: rmudgett
JIRA SWP-1192
(closes issue 0018395)
Reported by: shihchuan
Tested by: rmudgett
(closes issue 0017273)
Reported by: grecco
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1047/ [^]
------------------------------------------------------------------------
------------------------------------------------------------------------
r303106 | sruffell | 2011-01-20 13:56:35 -0600 (Thu, 20 Jan 2011) | 15
lines
main/features: Use POLLPRI when waiting for events on parked channels.
This change resolves a regression in the 1.6.2 when converting from
select to poll. The DAHDI timers use POLLPRI to indicate that the
timer
fired, but features was not waiting for that flag. The result was no
audio for MOH when a call was parked and res_timing_dahdi was in use.
This patch is slightly modified from the one on the mantis issue. It
does
not set an exception on the channel if the POLLPRI flag is set.
(closes issue 0018262)
Reported by: francesco_r
Patches:
patch_park_moh-trunk-2.txt uploaded by cjacobsen (license 1029)
Tested by: francesco_r, rfrantik, one47
------------------------------------------------------------------------
2011-01-14 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.17-rc1 Released.
2011-01-14 20:03 +0000 [r301842-301848] lathama <lathama@localhost>:
* funcs/func_base64.c, funcs/func_aes.c: Add relationships to
function documentation. Fix amatuer type mistake
* funcs/func_base64.c, funcs/func_aes.c: Add relationships to
function documentation.
2011-01-13 17:01 +0000 [r301730] Leif Madsen <lmadsen@digium.com>
* configs/phoneprov.conf.sample: Add static entry for split Polycom
332 firmware. (closes issue #18607) Reported by: cjacobsen
Patches: polycom_331.diff uploaded by cjacobsen (license 1029)
Tested by: lathama
2011-01-12 21:05 +0000 [r301682] Terry Wilson <twilson@digium.com>
* channels/chan_sip.c: Don't reject all SUBSCRIBE auth requests
When merging another SUBSCRIBE fix from 1.4, some braces were put
in the wrong place. This patch fixes that. (closes issue #18597)
Reported by: thsgmbh
2011-01-12 18:50 +0000 [r301594] Matthew Nicholson <mnicholson@digium.com>
* main/manager.c, /: Removed a usleep(1) that shouldn't be
necessary in session_do, and removed the ms_t member from the
mansession_session structure. Merged revisions 301591 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r301591 | mnicholson | 2011-01-12 12:39:03 -0600 (Wed, 12 Jan
2011) | 5 lines Don't store the thread id for the manager session
in the structure we pass to the thread for the manager session.
ABE-2543 ........
2011-01-12 18:11 +0000 [r301503] Jeff Peeler <jpeeler@digium.com>
* main/channel.c, /: Merged revisions 301502 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r301502 | jpeeler | 2011-01-12 12:10:42 -0600 (Wed, 12 Jan 2011)
| 12 lines Fix CPU spike when pressing DTMF after agent login.
The problem here is that DTMF was being continuously deferred and
requeued since ast_safe_sleep is called in a loop. There are
serveral other places in the code that sleeps and then loops in a
similar fashion. Because of this fact I opted to not defer DTMF
any more, which will not affect the original fix:
https://reviewboard.asterisk.org/r/674 (closes issue #18130)
Reported by: rgj ........
2011-01-11 19:14 +0000 [r301310] Paul Belanger <pabelanger@digium.com>
* configs/extensions.conf.sample: Fix a logic issue when passing
context ARG
2011-01-11 18:42 +0000 [r301307] Matthew Nicholson <mnicholson@digium.com>
* /, main/utils.c: Merged revisions 301305 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r301305 | mnicholson | 2011-01-11 12:34:40 -0600 (Tue, 11 Jan
2011) | 4 lines Prevent buffer overflows in ast_uri_encode()
ABE-2705 ........
2011-01-09 21:38 +0000 [r301176-301220] Paul Belanger <pabelanger@digium.com>
* autoconf/ast_ext_lib.m4, configure, configure.ac: SOUND_CACHE_DIR
now defaults to empty Sounds files included in the Asterisk
tarball were being ignored and re-downloaded. Users wanting to
cache the files can still override the setting using the
--with-sounds-cache option. (closes issue #18589) Reported by:
pabelanger Patches: issue18589.patch uploaded by pabelanger
(license 224) Tested by: pabelanger Review:
https://reviewboard.asterisk.org/r/1074/
* apps/app_verbose.c: Indicate log level argument for Log() is not
optional (closes issue #18586) Reported by: kshumard Patches:
app_verbose.c.patch uploaded by kshumard (license 92)
2011-01-07 20:52 +0000 [r301089] Jason Parker <jparker@digium.com>
* apps/app_meetme.c: Initialize useropts/adminopts in case there is
no column in the realtime DB. (closes issue #18182) Reported by:
dimas Patches: v1-18182.patch uploaded by dimas (license 88)
Tested by: dimas
2011-01-07 19:57 +0000 [r300951-301046] Jeff Peeler <jpeeler@digium.com>
* apps/app_voicemail.c: Fix regression causing forwarding
voicemails to not work with file storage. I had actually already
fixed this in 295200 in 1.4 and thought it wasn't missing in the
other branches for some reason. (closes issue #18358) Reported
by: cabal95
* apps/app_voicemail.c, /: Merged revisions 300918 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r300918 | jpeeler | 2011-01-07 11:13:21 -0600 (Fri, 07
Jan 2011) | 7 lines Ensure good bye prompt in voicemail is played
at the correct time. Specifically in the case of timing out but
not leaving voicemail nothing should be heard. And when leaving
voicemail it should be heard. ABE-2647 ........
2011-01-05 18:54 +0000 [r300622] Tilghman Lesher <tilghman@meg.abyt.es>
* res/res_odbc.c, /: Merged revisions 300621 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r300621 | tilghman | 2011-01-05 12:47:46 -0600 (Wed, 05 Jan 2011)
| 10 lines Use the sanity check in place of the
disconnect/connect cycle. The disconnect/connect cycle has the
potential to cause random crashes. (closes issue #18243) Reported
by: ks3 Patches: res_odbc.patch uploaded by ks3 (license 1147)
Tested by: ks3 ........
2011-01-05 16:28 +0000 [r300574] Paul Belanger <pabelanger@digium.com>
* cdr/cdr_sqlite.c: Change deprecated message to LOG_WARNING Also
removed latter part of message Discussed on #asterisk-dev
2011-01-04 21:52 +0000 [r300431-300520] Leif Madsen <lmadsen@digium.com>
* channels/chan_iax2.c, main/xmldoc.c, channels/chan_sip.c,
channels/chan_agent.c: Fix backwards and broken XML
documentation. (closes issue #18547) Reported by: jcovert
Patches: xmldoc.c.patch uploaded by jcovert (license 551)
chan_iax2.c.doc.patch uploaded by jcovert (license 551)
chan_sip.c.patch uploaded by jcovert (license 551)
chan_agent.c.patch uploaded by jcovert (license 551)
* configs/users.conf.sample: Add some documentation to
users.conf.sample. (closes issue #18531) Reported by: lathama
Patches: users.conf.sample2.diff uploaded by lathama (license
1028) Tested by: lathama
2011-01-04 20:59 +0000 [r300429] Russell Bryant <russell@digium.com>
* contrib/scripts/autosupport, /, contrib/scripts/autosupport.8:
Merged revisions 300428 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r300428 | russell | 2011-01-04 14:56:04 -0600 (Tue, 04 Jan 2011)
| 4 lines Update the autosupport script from Digium support.
(closes AST-395) ........
2011-01-04 17:37 +0000 [r300298] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c: Merged revisions 300216 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r300216 | twilson | 2011-01-04 11:11:48 -0600 (Tue, 04 Jan 2011)
| 15 lines Don't authenticate SUBSCRIBE re-transmissions This
only skips authentication on retransmissions that are already
authenticated. A similar method is already used for INVITES. This
is the kind of thing we end up having to do when we don't have a
transaction layer... (closes issue #18075) Reported by: mdu113
Patches: diff.txt uploaded by twilson (license 396) Tested by:
twilson, mdu113 Review: https://reviewboard.asterisk.org/r/1005/
........
2011-01-03 23:02 +0000 [r300165] Richard Mudgett <rmudgett@digium.com>
* main/features.c: Use correct variable for atxfercallbackretries
config option. * Misc formatting changes.
2010-12-28 18:51 +0000 [r299864] Paul Belanger <pabelanger@digium.com>
* apps/app_chanspy.c: Documentation typo
2010-12-25 10:05 +0000 [r299625] Tilghman Lesher <tilghman@meg.abyt.es>
* channels/chan_local.c, /: Merged revisions 299624 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r299624 | tilghman | 2010-12-25 04:04:06 -0600 (Sat, 25
Dec 2010) | 5 lines Move check for extension existence below
variable inheritance, due to the possible use of an eswitch.
(closes issue #16228) Reported by: jlaguilar ........
2010-12-23 03:02 +0000 [r299530-299533] Moises Silva <moises.silva@gmail.com>
* channels/chan_dahdi.c: do not use progress which is for PRI and
SS7, add mfcr2_progress member
* channels/chan_dahdi.c: Enqueue AST_CONTROL_PROGRESS after
AST_CONTROL_RINGING when MFC-R2 calls are accepted (closes issue
#18438) Reported by: mariner7 Tested by: moy
2010-12-22 20:03 +0000 [r299448] Tilghman Lesher <tilghman@meg.abyt.es>
* pbx/ael/ael-test/ref.ael-test19,
pbx/ael/ael-test/ref.ael-vtest13, res/ael/pval.c,
pbx/ael/ael-test/ref.ael-vtest25,
pbx/ael/ael-test/ref.ael-vtest17, pbx/ael/ael-test/ref.ael-test3:
Resolve warnings by disambiguating the "s" extension as used by
chan_dahdi from the "s" extension as used by the AEL macros.
(closes issue #18480) Reported by: nivek Patches:
20101215__issue18480__2.diff.txt uploaded by tilghman (license
14) Tested by: nivek
2010-12-20 21:25 +0000 [r299242] Matthew Nicholson <mnicholson@digium.com>
* /, channels/chan_sip.c: Merged revisions 299194,299198,299220 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r299194 | mnicholson | 2010-12-20 14:45:38 -0600 (Mon, 20 Dec
2010) | 6 lines Respond as soon as possible with a 202 Accepted
to refer requests. This change also plugs a few memory leaks that
can occur when parking sip calls. ABE-2656 ........ r299198 |
mnicholson | 2010-12-20 15:00:44 -0600 (Mon, 20 Dec 2010) | 2
lines Remove changes to via processing that were not supposed to
go into the last commit. ........ r299220 | mnicholson |
2010-12-20 15:21:39 -0600 (Mon, 20 Dec 2010) | 4 lines Use
ast_free() instead of free() ABE-2656 ........
2010-12-20 18:16 +0000 [r299130-299136] Tilghman Lesher <tilghman@meg.abyt.es>
* sample.call: Documentation fix
* cdr/cdr_pgsql.c: If a call was not answered, then the billsec was
calculated unusually large. Also, due to a copy and paste error,
a request for the answer field would have given the start value,
instead. (closes issue #18460) Reported by: joscas Patches:
20101215__issue18460.diff.txt uploaded by tilghman (license 14)
Tested by: joscas
2010-12-20 16:18 +0000 [r299087] Leif Madsen <lmadsen@digium.com>
* main/features.c: Note that Park() timeout is milliseconds.
(closes issue #15758) Reported by: mmurdock Tested by: mmurdock,
seanbright
2010-12-20 09:13 +0000 [r299003] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* channels/chan_sip.c: Typos: recieved => received
2010-12-18 00:08 +0000 [r298817-298962] Tilghman Lesher <tilghman@meg.abyt.es>
* main/say.c: Remove backtrace used for testing merge process
* main/astobj2.c, utils/conf2ael.c, include/asterisk/logger.h,
configure, build_tools/menuselect-deps.in, main/logger.c,
utils/ael_main.c, utils/hashtest2.c, makeopts.in,
utils/check_expr.c, utils/refcounter.c, include/asterisk/utils.h,
build_tools/cflags-devmode.xml, /, main/Makefile,
include/asterisk/autoconfig.h.in, main/say.c, configure.ac,
utils/hashtest.c, main/utils.c: Merged revisions 298905 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r298905 | tilghman | 2010-12-17 15:40:56 -0600 (Fri, 17 Dec 2010)
| 6 lines Let Asterisk find better backtrace information with
libbfd. The menuselect option BETTER_BACKTRACES, if enabled, will
use libbfd to search for better symbol information within both
the Asterisk binary, as well as loaded modules, to assist when
using inline backtraces to track down problems. ........
* configure, configure.ac: Also include PTHREAD_LIBS and
PTHREAD_CFLAGS for SQLite 3, as it's needed on some platforms.
(closes issue #18493) Reported by: pprindeville Patches:
asterisk-1.8-sqlite3.patch uploaded by pprindeville (license 347)
Tested by: pprindeville
2010-12-16 23:30 +0000 [r298597-298684] Jeff Peeler <jpeeler@digium.com>
* apps/app_voicemail.c, /: Merged revisions 298683 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r298683 | jpeeler | 2010-12-16 17:29:30 -0600 (Thu, 16
Dec 2010) | 2 lines After recording only silence for a voicemail
prepending, restore backup files. ........
* apps/app_queue.c, /: Merged revisions 298596 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r298596 | jpeeler | 2010-12-16 14:46:52 -0600 (Thu, 16 Dec 2010)
| 7 lines Fix improper hangup when doing an attended transfer to
queue. Had to indicate ringing in wait_for_answer so the attended
transfer code would not try and hang up the local channel it
created, which would kill the call. ABE-2624 ........
2010-12-16 09:04 +0000 [r298393-298481] Tilghman Lesher <tilghman@meg.abyt.es>
* res/res_config_odbc.c, /: Merged revisions 298480 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r298480 | tilghman | 2010-12-16 03:03:40 -0600 (Thu, 16
Dec 2010) | 14 lines Only increment the pointer once per loop,
otherwise we corrupt the value. (closes issue #18251) Reported
by: bcnit Patches: 20101110__issue18251.diff.txt uploaded by
tilghman (license 14) Tested by: trev, jthurman, elguero (closes
issue #18279) Reported by: zerohalo Patches:
20101109__issue18279.diff.txt uploaded by tilghman (license 14)
Tested by: zerohalo ........
* funcs/func_dialgroup.c: Eliminate duplicates from container.
(closes issue #18091) Reported by: bunny Patches:
20101006__issue18091.diff.txt uploaded by tilghman (license 14)
Tested by: bunny
* /, cdr/cdr_sqlite.c: Merged revisions 298392 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r298392 | tilghman | 2010-12-15 18:28:04 -0600 (Wed, 15 Dec 2010)
| 8 lines Unregister before shutting down the connection, to
avoid a race. (closes issue #18481) Reported by: pabelanger
Patches: 20101215__issue18481.diff.txt uploaded by tilghman
(license 14) Tested by: pabelanger ........
2010-12-15 21:31 +0000 [r298346] Sean Bright <sean@malleable.com>
* main/astobj2.c, /: Merged revisions 298345 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r298345 | seanbright | 2010-12-15 16:28:29 -0500 (Wed, 15 Dec
2010) | 6 lines Fix reference and container leaks when running
'astobj2 test.' We need to make sure that ao2_iterator_destroy is
called once for each time that ao2_iterator_init is called. Also
make sure to unref a newly allocated object that we've linked
into a container. ........
2010-12-13 17:04 +0000 [r298194] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 298193 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r298193 | rmudgett | 2010-12-13 10:56:07 -0600 (Mon, 13
Dec 2010) | 19 lines Outgoing PRI/BRI calls cannot do DTMF
triggered transfers. Outgoing PRI/BRI calls cannot do DTMF
triggered transfers if a PROCEEDING message is not received. The
debug output shows that the DTMF begin event is seen, but the
DTMF end event is missing. When the DTMF begin happens, the call
is muted so we now have one way audio (until a DTMF end event is
somehow seen). * Made set the proceeding flag when the
PRI_EVENT_ANSWER event is received. * Made absorb the DTMF begin
and DTMF end events if we are overlap dialing and have not seen a
PROCEEDING message. * Added a debug message when absorbing a DTMF
event. JIRA SWP-2690 JIRA ABE-2697 ........
2011-01-12 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.16 Released.
2011-01-12 Leif Madsen <lmadsen@digium.com>
* Merge in changes for configure script to resolve issue for
Debian package builders.
------------------------------------------------------------------------
r301220 | pabelanger | 2011-01-09 15:38:25 -0600 (Sun, 09 Jan 2011)
| 14 lines
SOUND_CACHE_DIR now defaults to empty
Sounds files included in the Asterisk tarball were being ignored and
re-downloaded. Users wanting to cache the files can still override
the setting
using the --with-sounds-cache option.
(closes issue 0018589)
Reported by: pabelanger
Patches:
issue18589.patch uploaded by pabelanger (license 224)
Tested by: pabelanger
Review: https://reviewboard.asterisk.org/r/1074/ [^]
------------------------------------------------------------------------
2010-12-13 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.16-rc1 Released.
2010-12-10 16:24 +0000 [r298050] Tilghman Lesher <tlesher@digium.com>
* main/netsock.c, configure, include/asterisk/autoconfig.h.in,
configure.ac: Portability issue on OpenSolaris. Also detect the
required structure element, because OpenSolaris defines
SIOCGIFHWADDR, but without support for IP sockets. (closes issue
#18442) Reported by: ranjtech Patches:
20101209__issue18442.diff.txt uploaded by tilghman (license 14)
Tested by: ranjtech
2010-12-09 22:10 +0000 [r297960] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c: Merged revisions 297959 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r297959 | twilson | 2010-12-09 16:00:30 -0600 (Thu, 09 Dec 2010)
| 14 lines Ignore spurious REGISTER requests If a REGISTER
request with a Call-ID matching an existing transaction is
received it was possible that the REGISTER request would
overwrite the initreq of the private structure. This info is used
to generate messages for other responses in the transaction. This
patch ignores REGISTER requests that match non-REGISTER
transactions. (closes issue #18051) Reported by: eeman Tested by:
twilson Review: https://reviewboard.asterisk.org/r/1050/ ........
2010-12-08 18:04 +0000 [r297908] Tilghman Lesher <tlesher@digium.com>
* configs/extensions.conf.sample: Use inheritance to get correct
results for SIPFROMDOMAIN. (from an internal Digium discussion)
2010-12-07 22:58 +0000 [r297824] Jeff Peeler <jpeeler@digium.com>
* main/channel.c, /: Merged revisions 297823 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r297823 | jpeeler | 2010-12-07 16:57:48 -0600 (Tue, 07 Dec 2010)
| 12 lines Revert code that changed SSRC for DTMF. Some previous
behavior was attempted to be restored, but mistakingly I did not
realize that the previous behavior was incorrect. This fixes DTMF
not being detected since DTMF shouldn't cause the SSRC to change.
(related to issue #17404) (closes issue #18189) (closes issue
#18352) Reported by: marcbou Tested by: cmbaker82 ........
2010-12-07 22:40 +0000 [r297713-297819] Tilghman Lesher <tlesher@digium.com>
* contrib/init.d/org.asterisk.muted.plist (added), Makefile,
utils/muted.c, /: Merged revisions 297818 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r297818 | tilghman | 2010-12-07 16:35:50 -0600 (Tue, 07 Dec 2010)
| 4 lines Use non-deprecated APIs for CoreAudio Review:
https://reviewboard.asterisk.org/r/1040/ ........
* apps/app_followme.c, /: Merged revisions 297689 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r297689 | tilghman | 2010-12-06 18:07:37 -0600 (Mon, 06 Dec 2010)
| 8 lines Don't create a Local channel if the target extension
does not exist. (closes issue #18126) Reported by: junky Patches:
followme.diff uploaded by junky (license 177) (partially
restructured by me to avoid a possible memory leak) ........
2010-12-06 22:03 +0000 [r297605] Jeff Peeler <jpeeler@digium.com>
* /, channels/chan_sip.c: Merged revisions 297603 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r297603 | jpeeler | 2010-12-06 15:57:15 -0600 (Mon, 06 Dec 2010)
| 12 lines Improve handling of REGISTER requests with multiple
contact headers. The changes here attempt to more strictly follow
RFC 3261 section 10.3. Basically the following will now cause a
400 Bad Response to be returned, if: - multiple Contact headers
are present with one set to expire all bindings ("*") - wildcard
parameter is specified for Contact without Expires header or
Expires header is not set to zero. ABE-2442 ABE-2443 ........
2010-12-03 17:40 +0000 [r297534] Sean Bright <sean@malleable.com>
* channels/chan_console.c: The CLI command should not contain
<placeholder>s, these are for descriptions.
2010-12-02 20:06 +0000 [r297405] Paul Belanger <pabelanger@digium.com>
* Makefile, /: Merged revisions 297404 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r297404 | pabelanger | 2010-12-02 15:01:08 -0500 (Thu, 02 Dec
2010) | 7 lines Resolve compile error under FreeBSD We now set
_ASTCFLAGS+=-march=i686 for i386 processors, still allowing
ASTCFLAGS to override the setting. Review:
https://reviewboard.asterisk.org/r/1043/ ........
2010-12-02 18:07 +0000 [r297311] Terry Wilson <twilson@digium.com>
* /, main/abstract_jb.c: Merged revisions 297310 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r297310 | twilson | 2010-12-02 12:00:27 -0600 (Thu, 02 Dec 2010)
| 12 lines Initialize offset for adaptive jitter buffer When the
adaptive jitter buffer is enabled in sip.conf, the first frame
placed in the jitter buffer fails with something like:
jb_warning_output: Resyncing the jb. last_delay 0, this delay
-215886466, threshold 1000, new offset 215886466 This happens
because the offset is not initialized before calling jb_put().
This patch modifies jb_put_first_adaptive() to set the offset to
the frame's timestamp. Review:
https://reviewboard.asterisk.org/r/1041/ ........
2010-12-02 13:16 +0000 [r297229] Russell Bryant <russell@digium.com>
* /, apps/app_meetme.c: Merged revisions 297228 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r297228 | russell | 2010-12-02 07:16:15 -0600 (Thu, 02 Dec 2010)
| 6 lines Add "DAHDI" to a couple of app_meetme error messages.
This is in response to some questions on IRC. To the user, there
was nothing that made it obvious that this error had anything to
do with DAHDI not being loaded. ........
2010-12-02 08:55 +0000 [r297186] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Merged revisions 297185 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r297185 | oej | 2010-12-02 09:37:17 +0100 (Tor, 02 Dec 2010) | 5
lines If we get a NOTIFY from a non-existing subscription we
should answer with 481, not bad event. If we answer 481 the
subscription that we don't want will be cancelled. ........
2010-12-01 17:52 +0000 [r297073] Jeff Peeler <jpeeler@digium.com>
* /, channels/chan_sip.c: Merged revisions 297072 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r297072 | jpeeler | 2010-12-01 11:50:09 -0600 (Wed, 01 Dec 2010)
| 23 lines Fix not stopping MOH when transfered local channel
queue member is answered. The problem here is only present when
local channels are used with the MOH passthru option as well as
no optimization (/nm). I will describe the slightly bizarre
scenario that was used to test, where phones B and C are queue
members: Phone A dials into a queue with two members using local
channels and the above options. Phone B answers. Phone A blind
transfers phone B into the same queue. Phone A hangs up. Phone C
answers, but phone B didn't stop playing MOH. In this scenario,
the unhold frame that should have gotten to phone B never arrived
due to the masquerade from the blind transfer. This is usually
fine since app_queue manages the starting and stopping of MOH.
However, with the passthrough option enabled when app_queue
attempts to stop MOH it tries to do so on the local channel
rather than the real channel. The easiest solution was to just
make sure to send an unhold frame during the transfer since it
wouldn't make sense to have MOH playing after a transfer anyway.
This only modifies SIP transfers, but the other transfers did not
seem to be a problem. If DTMF based transfers were a problem it
might be okay to add ast_moh_stop to finishup, but I didn't want
to have to add that unless required. ABE-2624 ........
2010-12-01 17:01 +0000 [r296950-296991] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/frame.h, /: Merged revisions 296990 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r296990 | tilghman | 2010-12-01 10:59:26 -0600 (Wed, 01
Dec 2010) | 5 lines Clarify documentation on how we store codec
preference lists. (closes issue #18397) Reported by: birgita
........
* channels/chan_iax2.c: Missed initializations caused startup
errors on Mac OS X (and possibly others, too).
2010-12-01 00:24 +0000 [r296869] Jeff Peeler <jpeeler@digium.com>
* apps/app_voicemail.c, /: Merged revisions 296868 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r296868 | jpeeler | 2010-11-30 18:23:19 -0600 (Tue, 30
Nov 2010) | 4 lines Properly restore backup information file when
hanging up during message prepending. ABE-2654 ........
2010-11-29 22:54 +0000 [r296671] Paul Belanger <pabelanger@digium.com>
* channels/chan_iax2.c, /: Merged revisions 296670 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r296670 | pabelanger | 2010-11-29 17:49:39 -0500 (Mon,
29 Nov 2010) | 5 lines Make sure nothing else is needed before
destroying the scheduler. (closes issue #18398) Reported by:
pabelanger ........
2010-11-29 07:27 +0000 [r296533] Tilghman Lesher <tlesher@digium.com>
* main/asterisk.c, configure, include/asterisk/autoconfig.h.in,
configure.ac: I love standards. There are so many to choose from.
Except when there isn't one. Linux and *BSD disagree on the
elements within the ucred structure. Detect which one is in use
on the system. (closes issue #18384) Reported by: bjm Patches:
cred-diffs uploaded by bjm (license 473)
20101127__issue18384__1.6.2.diff.txt uploaded by tilghman
(license 14) 20101127__issue18384__1.8.diff.txt uploaded by
tilghman (license 14) Tested by: tilghman, bjm
2010-11-27 10:39 +0000 [r296466] Tilghman Lesher <tlesher@digium.com>
* apps/app_meetme.c: 18 characters is too short for most date/times
(20 is the usual, but we add more in case of greater precision).
(closes issue #18369) Reported by: tnakonz
2010-11-26 12:23 +0000 [r296351] Olle Johansson <oej@edvina.net>
* /, main/say.c: Merged revisions 296309 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r296309 | oej | 2010-11-26 10:53:31 +0100 (Fre, 26 Nov 2010) | 11
lines Fix bugs in saying numbers using the Swedish language
syntax (closes issue #18355) Reported by: oej Patch by: oej Much
help from Peter Lindahl. Testing by the ClearIT team during a
coffee break. Review: https://reviewboard.asterisk.org/r/1033/
........
2010-11-24 23:28 +0000 [r296221] Russell Bryant <russell@digium.com>
* main/channel.c, /: Merged revisions 296213 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r296213 | russell | 2010-11-24 17:26:43 -0600 (Wed, 24 Nov 2010)
| 6 lines Make Asterisk less crashy. Since we might not put a new
translation path on the channel, go ahead and set it to NULL
right after destroying the old one to ensure we don't try to free
an invalid translation path later on. ........
2010-11-24 22:42 +0000 [r296166] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 296165 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r296165 | rmudgett | 2010-11-24 16:41:07 -0600 (Wed, 24
Nov 2010) | 43 lines Oneway audio to SIP phone from FXS port
after FXS port gets a CallWaiting pip. The FXS connected phone
has to have CW/CID support to fail, as it will send back a DTMF
'A' or 'D' when it's ready to receive CallerID. A normal phone
with no CID never fails. Also the SIP phone does not hear MOH
when the CW call is answered. The DTMF end frame is suppressed
when the phone acknowledges the CW signal for CID. The problem is
the DTMF begin frame needs to be suppressed as well. The DTMF
begin frame is causing SIP to start sending the DTMF RTP frames.
Since the DTMF end frame is suppressed, SIP will not stop sending
those DTMF RTP packets. * Suppress the DTMF begin and end frames
when the channel driver is looking for DTMF digits. * Fixed a
couple issues caused by not cleaning up the CID spill if you
answer the CW call while it is sending the CID spill. * Fixed not
sending CW/CID spill to the phone when the call is natively
bridged. (Fixed by not using native bridge if CW/CID is
possible.) * Suppress received audio when sending CW/CID spills.
The other parties involved do not need to hear the CW/CID spills
and may be confused if the CW call is for them. (closes issue
#18129) Reported by: alecdavis Patches: issue_18129_v1.8_v3.patch
uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett
NOTE: * v1.4 does not have the main problem fixed by suppressing
the DTMF start frames. The other three items fixed are relevant.
* If you really must restore native bridging between analog
ports, you need to disable CW/CID either by configuring
chan_dahdi.conf callwaitingcallerid=no or dialing *70 before
dialing the number to temporarily disable CW. ........
2010-11-24 20:23 +0000 [r296001-296083] Russell Bryant <russell@digium.com>
* main/channel.c, /: Merged revisions 296082 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r296082 | russell | 2010-11-24 14:22:32 -0600 (Wed, 24 Nov 2010)
| 12 lines Fix false reporting of an error by set_format(). In
the case that the native format was able to be changed to match
the new requested format, the code proceeded to attempt to build
a translation path, anyway. The result would be NULL, since no
translation path is necessary and resulted in this function
thinking an error has occurred. This case is now specifically
caught and no attempt to build a translation path is attempted.
Thanks to our automated tests and bamboo.asterisk.org for
catching this problem and making a whole lot of noise when things
started failing. :-) ........
* apps/app_dial.c, main/channel.c, /: Merged revisions 296000 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r296000 | russell | 2010-11-24 10:48:39 -0600 (Wed, 24 Nov 2010)
| 38 lines Handle failures building translation paths more
effectively. The problem scenario occurred on a heavily loaded
system that was using the codec_dahdi module and exceeded the
hardware transcoding capacity. The failure mode at that point was
not good. The report came in to us as an Asterisk lock-up. The
"core show locks" shows a ton of threads locked up (but no
obvious deadlock). Upon deeper investigation, when the system is
in this state, the CPU was maxed out. The CPU was being consumed
by the Asterisk logger spewing messages on every audio frame for
calls set up after transcoder capacity was reached. The purpose
of this patch is to make Asterisk handle failures to create a
translation path in a more graceful manner. If we can't
translate, then the call just needs to be dropped, as it's not
going to work. These are the changes: 1) In set_format() of
channel.c (which is called by set_read_format() and
set_write_format()), it was ignoring if
ast_translator_build_path() failed and returned NULL. It now pays
attention to that case and returns a result reflecting failure.
With this change in place, the bridging code will immediately
detect a failure and end the bridge instead of proceeding to try
to bridge frames that can't be translated and making channel
drivers freak out by sending them frames in a format they weren't
expecting. 2) In ast_indicate_data() of channel.c, failure of
ast_playtones_start() was ignored. It is now reflected in the
return value of the function. This didn't turn out to have any
affect on the bug, but seemed like a good change to leave in. 3)
In app_dial(), when only sending a call to a single endpoint, it
will attempt to do some bridging of its own of early audio. It
uses make_compatible() when it's going to do this. However, it
ignored failure from make compatible. So, even with the fix from
#1, if there was early audio going through app_dial, there would
still be a period of invalid frames passing through. After
detecting failure here, Dial() exits. ABE-2658 ........
2010-11-23 09:36 +0000 [r295907] Olle Johansson <oej@edvina.net>
* /, main/say.c: Merged revisions 295906 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r295906 | oej | 2010-11-23 10:28:14 +0100 (Tis, 23 Nov 2010) | 8
lines Fix support of saynumber(1,n) in the Swedish language
(closes issue #18353) Reported by: oej Review:
https://reviewboard.asterisk.org/r/1031/ ........
2010-11-22 20:02 +0000 [r295868] Sean Bright <sean@malleable.com>
* configs/chan_dahdi.conf.sample: Change some documentation to
suggest dahdi_monitor instead of ztmonitor.
2010-11-22 19:28 +0000 [r295843] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/frame.h, main/channel.c, main/pbx.c, /,
apps/app_macro.c, include/asterisk/channel.h: Merged revisions
295790 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r295790 | rmudgett | 2010-11-22 12:46:26 -0600 (Mon, 22 Nov 2010)
| 46 lines The channel redirect function (CLI or AMI) hangs up
the call instead of redirecting the call. To recreate the
problem: 1) Party A calls Party B 2) Invoke CLI "channel
redirect" command to redirect channel call leg associated with A.
3) All associated channels are hung up. Note that if the CLI
command were done on the channel call leg associated with B it
works. This regression was a result of the fix for issue #16946
(https://reviewboard.asterisk.org/r/740/). The regression affects
all features that use an async goto to execute the dialplan
because of an external event: Channel redirect, AMI redirect, SIP
REFER, and FAX detection. The struct ast_channel._softhangup code
is a mess. The variable is used for several purposes that do not
necessarily result in the call being hung up. I have added
doxygen comments to describe how the various _softhangup bits are
used. I have corrected all the places where the variable was
tested in a non-bit oriented manner. The primary fix is the new
AST_CONTROL_END_OF_Q frame. It acts as a weak hangup request so
the soft hangup requests that do not normally result in a hangup
do not hangup. JIRA SWP-2470 JIRA SWP-2489 (closes issue #18171)
Reported by: SantaFox (closes issue #18185) Reported by:
kwemheuer (closes issue #18211) Reported by: zahir_koradia
(closes issue #18230) Reported by: vmarrone (closes issue #18299)
Reported by: mbrevda (closes issue #18322) Reported by: nerbos
Review: https://reviewboard.asterisk.org/r/1013/ ........
2010-11-20 00:45 +0000 [r295710] Russell Bryant <russell@digium.com>
* include/asterisk/event.h, main/event.c: Fix cache of device state
changes for multiple servers. This patch addresses a regression
where device states across multiple servers were not being
processing completely correctly. The code works to determine the
overall state by looking at the last known state of a device on
each server. However, there was a regression due to some invasive
rewrites of how the cache works that led to the cache only
storing the last device state change for a device, regardless of
which server it was on. The code is set up to cache device state
change events by ensuring that each event in the cache has a
unique device name + entity ID (server ID). The code that was
responsible for comparing raw information elements (which EID is)
always returned a match due to a memcmp() with a length of 0.
There isn't much code to fix the actual bug. This patch also
introduces a new CLI command that was very useful for debugging
this problem. The command allows you to dump the contents of the
event cache. (closes issue #18284) Reported by: klaus3000
Patches: issue18284.rev1.txt uploaded by russell (license 2)
Tested by: russell, klaus3000 (closes issue #18280) Reported by:
klaus3000 Review: https://reviewboard.asterisk.org/r/1012/
2010-11-19 21:55 +0000 [r295672] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c: Merged revisions 295628 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r295628 | twilson | 2010-11-19 12:53:36 -0800 (Fri, 19 Nov 2010)
| 8 lines Discard responses with more than one Via This is not a
perfect solution as headers that are joined via commas are not
detected. This is a parsing issue that to fix "correctly" would
necessitate a new SIP parser. Review:
https://reviewboard.asterisk.org/r/1019/ ........
2010-11-18 17:51 +0000 [r295440] Paul Belanger <pabelanger@digium.com>
* res/res_jabber.c, include/asterisk/jabber.h: Fix compiler
warnings when using openssl-dev 1.0.0+ Review:
https://reviewboard.asterisk.org/r/1016/
2010-11-16 22:57 +0000 [r295281] Richard Mudgett <rmudgett@digium.com>
* main/channel.c, /: Merged revisions 295280 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r295280 | rmudgett | 2010-11-16 16:52:06 -0600 (Tue, 16 Nov 2010)
| 1 line Dead code elimination in channel.c:ast_channel_bridge()
variable who. ........
2010-12-02 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.15 Released.
2010-11-15 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.15-rc1
2010-11-15 18:24 +0000 [r294988-295062] Tilghman Lesher <tlesher@digium.com>
* tests/test_expr.c (added), /: Merged revisions 295026 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r295026 | tilghman | 2010-11-15 11:58:37 -0600 (Mon, 15 Nov 2010)
| 2 lines Create test verifying results of expression parser
........
* funcs/func_curl.c: It is possible to crash Asterisk by feeding
the curl engine invalid data. (closes issue #18161) Reported by:
wdoekes Patches: 20101029__issue18161.diff.txt uploaded by
tilghman (license 14) Tested by: tilghman
2010-11-12 21:14 +0000 [r294904-294910] Jeff Peeler <jpeeler@digium.com>
* apps/app_voicemail.c: Return correct error code if lock path
fails. The recent changes to open_mailbox actually caused it to
be fixed, but let's be consistent. Reported by alecdavis in
asterisk-dev.
* apps/app_voicemail.c, /: Merged revisions 294903 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r294903 | jpeeler | 2010-11-12 14:49:09 -0600 (Fri, 12
Nov 2010) | 16 lines Fix regression causing abort in voicemail
after opening a mailbox with no mesgs. In order to be more safe,
some error handling code was changed to respect more error
conditions including the potential memory allocation failure for
deleted and heard message tracking introduced in 293004. However,
last_message_index returns -1 for zero messages (perhaps as
expected) and was triggering the stricter error checking. Because
last_message_index is only called directly in one place, just
return 0 from open_mailbox (for file based storage) when no
messages are detected unless a real error has occurred. (closes
issue #18240) Reported by: leobrown Patches:
bug18240.1-6-2.diff.txt uploaded by alecdavis (license 585)
Tested by: pabelanger ........
2010-11-12 02:44 +0000 [r294822] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 294821 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r294821 | rmudgett | 2010-11-11 20:41:13 -0600 (Thu, 11
Nov 2010) | 11 lines Asterisk is getting a "No D-channels
available!" warning message every 4 seconds. Asterisk is just
whining too much with this message: "No D-channels available!
Using Primary channel XXX as D-channel anyway!". Filtered the
message so it only comes out once if there is no D channel
available without an intervening D channel available period.
(closes issue #17270) Reported by: jmls ........
2010-11-11 21:57 +0000 [r294639-294733] Jeff Peeler <jpeeler@digium.com>
* /, channels/chan_sip.c: Merged revisions 294688 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r294688 | jpeeler | 2010-11-11 15:12:27 -0600 (Thu, 11 Nov 2010)
| 18 lines Fix problem with qualify option packets for realtime
peers never stopping. The option packets not only never stopped,
but if a realtime peer was not in the peer list multiple options
dialogs could accumulate over time. This scenario has the
potential to progress to the point of saturating a link just from
options packets. The fix was to ensure that the poke scheduler
checks to see if a peer is in the peer list before continuing to
poke. The reason a peer must be in the peer list to be able to
properly manage an options dialog is because otherwise the call
pointer is lost when the peer is regenerated from the database,
which is how existing qualify dialogs are detected. (closes issue
#16382) (closes issue #17779) Reported by: lftsy Patches:
bug16382-3.patch uploaded by jpeeler (license 325) Tested by:
zerohalo ........
* main/asterisk.c, include/asterisk.h, main/pbx.c, /: Merged
revisions 294384 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r294384 | jpeeler | 2010-11-09 11:37:59 -0600 (Tue, 09 Nov 2010)
| 47 lines Fix a deadlock in device state change processing.
Copied from some notes from the original author (Russell):
Deadlock scenario: Thread 1: device state change thread Holds -
rdlock on contexts Holds - hints lock Waiting on channels
container lock Thread 2: SIP monitor thread Holds the "iflock"
Holds a sip_pvt lock Holds channel container lock Waiting for a
channel lock Thread 3: A channel thread (chan_local in this case)
Holds 2 channel locks acquired within app_dial Holds a 3rd
channel lock it got inside of chan_local Holds a local_pvt lock
Waiting on a rdlock of the contexts lock A bunch of other threads
waiting on a wrlock of the contexts lock To address this
deadlock, some locking order rules must be put in place and
enforced. Existing relevant rules: 1) channel lock before a pvt
lock 2) contexts lock before hints lock 3) channels container
before a channel What's missing is some enforcement of the order
when you involve more than any two. To fix this problem, I put in
some code that ensures that (at least in the code paths involved
in this bug) the locks in (3) come before the locks in (2). To
change the operation of thread 1 to comply, I converted the
storage of hints to an astobj2 container. This allows processing
of hints without holding the hints container lock. So, in the
code path that led to thread 1's state, it no longer holds either
the contexts or hints lock while it attempts to lock the channels
container. (closes issue #18165) Reported by: antonio ABE-2583
........
2010-11-10 23:16 +0000 [r294571] Tilghman Lesher <tlesher@digium.com>
* main/features.c: Actually pay attention to documented settings in
features.conf. (closes issue #16757) Reported by: voxter Patches:
20101012__issue16757.diff.txt uploaded by tilghman (license 14)
Review: https://reviewboard.asterisk.org/r/994/
2010-11-10 12:41 +0000 [r294500] Russell Bryant <russell@digium.com>
* main/devicestate.c: Improve a debug message to be more readable
and consistent. (closes issue #18282) Reported by: klaus3000
Patches: ast_devstate2str-patch.txt uploaded by klaus3000
(license 65)
2010-11-09 20:27 +0000 [r294429] Tilghman Lesher <tlesher@digium.com>
* configure, configure.ac: Detect GMime properly on systems where
gmime flags and libs are configured with pkg-config. (closes
issue #16155) Reported by: jcollie Patches:
20100917__issue16155.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
2010-11-08 22:30 +0000 [r294277-294312] Jeff Peeler <jpeeler@digium.com>
* res/res_timing_timerfd.c: add missing unlock not present in
294277
* main/timing.c, main/channel.c, res/res_timing_timerfd.c,
include/asterisk/timing.h: Fix playback failure when using IAX
with the timerfd module. To fix this issue the alert pipe will
now be used when the timerfd module is in use. There appeared to
be a race that was not solved by adding locking in the timerfd
module, but needed to be there anyway. The race was between the
timer being put in non-continuous mode in ast_read on the channel
thread and the IAX frame scheduler queuing a frame which would
enable continuous mode before the non-continuous mode event was
read. This race for now is simply avoided. (closes issue #18110)
Reported by: tpanton Tested by: tpanton I put tested by tpanton
because it was tested on his hardware. Thanks for the remote
access to debug this issue!
2010-11-08 20:50 +0000 [r294242] Matthew Nicholson <mnicholson@digium.com>
* channels/chan_sip.c: Go off hold when we get an empty reinvite
telling us to. (closes issue 0014448) Reported by: frawd (closes
issue #17878) Reported by: frawd
2010-11-05 00:06 +0000 [r293969] Shaun Ruffell <sruffell@digium.com>
* codecs/codec_dahdi.c, /: Merged revisions 293968 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r293968 | sruffell | 2010-11-04 19:02:53 -0500 (Thu, 04
Nov 2010) | 17 lines codecs/codec_dahdi: Prevent "choppy" audio
when receiving unexpected frame sizes. dahdi-linux 2.4.0
(specifically commit 9034) added the capability for the wctc4xxp
to return more than a single packet of data in response to a
read. However, when decoding packets, codec_dahdi was still
assuming that the default number of samples was in each read. In
other words, each packet your provider sent you, regardless of
size, would result in 20 ms of decoded data (30 ms if decoding
G723). If your provider was sending 60 ms packets then
codec_dahdi would end up stripping 40 ms of data from each
transcoded frame resulting in "choppy" audio. This would only
affect systems where G729 packets are arriving in sizes greater
than 20ms or G723 packets arriving in sizes greater than 30ms.
DAHDI-744. ........
2010-11-03 18:31 +0000 [r293806] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 293805 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r293805 | rmudgett | 2010-11-03 13:23:04 -0500 (Wed, 03
Nov 2010) | 20 lines Party A in an analog 3-way call would
continue to hear ringback after party C answers. All parties are
analog FXS ports. 1) A calls B. 2) A flash hooks to call C. 3) A
flash hooks to bring C into 3-way call before C answers. (A and B
hear ringback) 4) C answers 5) A continues to hear ringback
during the 3-way call. (All parties can hear each other.) * Fixed
use of wrong variable in dahdi_bridge() that stopped ringback on
the wrong subchannel. * Made several debug messages have more
information. A similar issue happens if B and C are SIP channels.
B continues to hear ringback. For some reason this only affects
v1.8 and trunk. * Don't start ringback on the real and 3-way
subchannels when creating the 3-way conference. Removing this
code is benign on v1.6.2 and earlier. ........
2010-11-02 23:07 +0000 [r293723] Jeff Peeler <jpeeler@digium.com>
* /, channels/chan_sip.c: Merged revisions 293722 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r293722 | jpeeler | 2010-11-02 18:02:51 -0500 (Tue, 02 Nov 2010)
| 8 lines Add enabled/disabled information for rtautoclear sip
show settings output. When setting to zero/"no", the numeric
default was shown making it not obvious the disabled setting was
respected. (closes issue #18123) Reported by: zerohalo ........
2010-11-02 21:26 +0000 [r293647] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 293639 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r293639 | rmudgett | 2010-11-02 16:24:13 -0500 (Tue, 02
Nov 2010) | 6 lines Make warning message have more useful
information in it. Change "Unable to get index, and nullok is not
asserted" to "Unable to get index for '<channel-name>' on channel
<number> (<function>(), line <number>)". ........
2010-10-30 01:49 +0000 [r293340-293417] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 293416 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r293416 | rmudgett | 2010-10-29 20:45:49 -0500 (Fri, 29
Oct 2010) | 1 line Remove some more code that serves no purpose.
........
* channels/chan_dahdi.c, /: Merged revisions 293339 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r293339 | rmudgett | 2010-10-29 19:34:12 -0500 (Fri, 29
Oct 2010) | 1 line Remove some code that serves no purpose.
........
2010-10-28 19:54 +0000 [r293195-293196] Tilghman Lesher <tlesher@digium.com>
* main/ast_expr2.c, main/ast_expr2.h: Merged revisions 293194 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010)
| 5 lines "!00" evaluated as false, which is incorrect. Fixing.
Reported (though the reporter did not understand he was reporting
a bug) on the asterisk-users list:
http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html
........
* /, res/ael/ael.tab.c, main/ast_expr2.y, res/ael/ael_lex.c,
res/ael/ael.tab.h: Merged revisions 293194 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010)
| 5 lines "!00" evaluated as false, which is incorrect. Fixing.
Reported (though the reporter did not understand he was reporting
a bug) on the asterisk-users list:
http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html
........
2010-10-28 16:09 +0000 [r293158] Jeff Peeler <jpeeler@digium.com>
* funcs/func_strings.c: Fix infinite loop in FILTER(). Specifically
when you're using characters above \x7f or invalid character
escapes (e.g. \xgg). (closes issue #18060) Reported by: wdoekes
Patches: issue18060_func_strings_filter_infinite_loop.patch
uploaded by wdoekes (license 717) Tested by: wdoekes
2010-10-26 18:33 +0000 [r293118] Jeff Peeler <jpeeler@digium.com>
* apps/app_voicemail.c, /: Merged revisions 293004 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r293004 | jpeeler | 2010-10-25 17:55:28 -0500 (Mon, 25
Oct 2010) | 29 lines Fix inprocess_container in voicemail to
correctly restrict max messages. The comparison function logic
was off, so the number of sessions for a given mailbox were not
being incremented properly. This problem caused the maximum
number of messages per folder to not be respected when
simultaneously leaving multiple voicemails just below the
threshold. These problems should be fixed by the above, but just
in case: Fixed resequence_mailbox to rely on the actual number of
detected number of files in a directory rather than just assuming
only 10 messages more than the maximum had been left. Also if
more messages than the maximum are deleted they are actually
removed now. The second purpose of this commit should have been
separated out probably, but is related to the above. Again, if
the number of messages in a given voicemail folder exceeds the
maximum set limit make sure to allocate enough space for the
deleted and heard index tracking array. A few random fixes: There
was a forgotten decrement of the inprocess count in
imap_store_file. When using IMAP storage, do not look in the
directory where file based storage messages may still reside and
influence the message count. Ensure to use only the first format
in sendmail. ABE-2516 ........
2010-10-25 19:06 +0000 [r292867] David Vossel <dvossel@digium.com>
* channels/chan_local.c, /: Merged revisions 292866 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r292866 | dvossel | 2010-10-25 14:05:07 -0500 (Mon, 25
Oct 2010) | 27 lines This patch turns chan_local pvts into
astobj2 objects. chan_local does some dangerous things involving
deadlock avoidance. tech_pvt functions like hangup and
queue_frame are provided with a locked channel upon entry. Those
functions are completely safe as long as you don't attempt to
give up that channel lock, but that is impossible to guarantee
due to the required deadlock avoidance necessary to lock both the
tech_pvt and both channels involved. In the past, we have tried
to account for this by doing things like setting a "glare" flag
that indicates what function should destroy the pvt. This was
used in local_hangup and local_queue_frame to decided who should
destroy the pvt if they collided in separate threads. I have
removed the need to do this by converting all chan_local
tech_pvts to astobj2. This means we can ref a pvt before deadlock
avoidance and not have to worry about that pvt possibly getting
destroyed under us. It also cleans up where we destroy the
tech_pvt. The only unlink from the tech_pvt container occurs in
local_hangup now, which is where it should occur. Since there
still may be thread collisions on some functions like
local_hangup after deadlock avoidance, I have added some checks
to detect those collisions and exit appropriately. I think this
patch is going to solve quite a bit of weirdness we have had with
local channels in the past. ........
2010-10-22 21:16 +0000 [r292786] Leif Madsen <lmadsen@digium.com>
* contrib/scripts/asterisk.ldif, channels/chan_sip.c,
configs/res_ldap.conf.sample: Update the LDIF file for LDAP. The
LDIF file asterisk.ldif was quite a bit out of date from the
asterisk.ldap-schema file, so I've now updated that to be in
sync. The asterisk.ldif file being out of sync was a problem on
my systems where I was doing an ldapadd to import the schema into
the LDAP database, and the existing file would cause problems and
ERROR messages when registering. Additional documention has been
added based on feedback in the issue I'm closing. (closes issue
#13861) Reported by: scramatte Patches: ldap-update.txt uploaded
by lmadsen (license 10) Tested by: lmadsen, jcovert, suretec,
rgenthner
2010-10-21 13:11 +0000 [r292556] Leif Madsen <lmadsen@digium.com>
* configs/res_ldap.conf.sample: Change res_ldap.sample.conf to
match the schema. (closes issue #17376) Reported by: jcovert
Patches: res_ldap.conf.sample.patch uploaded by jcovert (license
551)
2010-10-21 00:05 +0000 [r292412] Paul Belanger <paul.belanger@polybeacon.com>
* apps/app_dial.c, /: Merged revisions 292411 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r292411 | pabelanger | 2010-10-20 20:00:51 -0400 (Wed, 20 Oct
2010) | 10 lines Record priv-recordintro as sln, not gsm This
removes the gsm->sln step when transcoding priv-recordintro.
(closes issue #18176) Reported by: pabelanger Patches:
chan_sip.diff uploaded by pabelanger (license 224) ........
2010-10-18 22:01 +0000 [r292229] Leif Madsen <lmadsen@digium.com>
* sounds/Makefile: Fix typo in the sounds/Makefile. (Issue #17426)
2010-10-18 21:54 +0000 [r292226] Jeff Peeler <jpeeler@digium.com>
* apps/app_voicemail.c, /: Merged revisions 292223 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r292223 | jpeeler | 2010-10-18 16:50:30 -0500 (Mon, 18
Oct 2010) | 11 lines Fix improper operator key acceptance and
clean up temp recording files. This is a fix for when pressing
the operator key after recording an unavailable, busy, name, or
temporary message in mailbox options. The operator key should not
be accepted here, but should be allowed during the message
recording. If the operator key is pressed during ensure the file
is saved or deleted as apporopriate. Also, ensure removal of
temporary recorded files after an early hang up or when message
acceptance confirmation times out. ABE-2518 ........
2010-10-18 21:50 +0000 [r292224] Leif Madsen <lmadsen@digium.com>
* sounds/Makefile, /, sounds/sounds.xml: Merged revisions 292222
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r292222 | lmadsen | 2010-10-18 16:47:25 -0500 (Mon, 18 Oct 2010)
| 9 lines Add support for the new English (Australian Accent)
sound files. (closes issue #17426) Reported by: camsown Patches:
core-sounds-en_AU.txt uploaded by camsown (license 1050)
add_AU_sounds.patch.txt uploaded by lmadsen (license 10) Tested
by: camsown, lmadsen, jtodd, qwell ........
2010-10-16 10:03 +0000 [r292049] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* res/res_musiconhold.c, configs/musiconhold.conf.sample: Base
directory for MOH should be ASTDATADIR If the directive
'directory' is relative, make it relative to the datadir, rather
than to the varlibdir. In the sample configuration it is relative
('moh'). This has no effect unless you have actively set the
datadir explicitly (at build time or at run time). (closes issue
#16906) Patches: moh_datadir uploaded by tzafrir (license 46)
Review: https://reviewboard.asterisk.org/r/974/
2010-10-15 19:35 +0000 [r291939] Paul Belanger <paul.belanger@polybeacon.com>
* configs/gtalk.conf.sample, /: Merged revisions 291938 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r291938 | pabelanger | 2010-10-15 15:30:41 -0400 (Fri, 15 Oct
2010) | 2 lines Clean up formatting. ........
2010-10-15 16:16 +0000 [r291904] Terry Wilson <twilson@digium.com>
* res/res_jabber.c: Don't crash or deadlock on module unload We
can't hold the lock while pthread_join is called since
aji_log_hook will attempt to lock from the other therad. We
reorder the pthread_join and ast_aji_disconnect so that we don't
do an SSL_read() while SSL_shutdown is running, causing a crash.
2010-10-13 23:36 +0000 [r291655] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 291643 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r291643 | rmudgett | 2010-10-13 18:29:58 -0500 (Wed, 13
Oct 2010) | 20 lines Deadlock between dahdi_exception() and
dahdi_indicate(). There is a deadlock between dahdi_exception()
and dahdi_indicate() for analog ports. The call-waiting and
three-way-calling feature can experience deadlock if these
features are trying to do something and an event from the bridged
channel happens at the same time. Deadlock avoidance code added
to obtain necessary channel locks before attemting an operation
with call-waiting and three-way-calling. (closes issue #16847)
Reported by: shin-shoryuken Patches: issue_16847_v1.4.patch
uploaded by rmudgett (license 664) issue_16847_v1.6.2.patch
uploaded by rmudgett (license 664) issue_16847_v1.8_v2.patch
uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett
Review: https://reviewboard.asterisk.org/r/971/ ........
2010-10-13 22:58 +0000 [r291580] Terry Wilson <twilson@digium.com>
* main/channel.c, /: Merged revisions 291577 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r291577 | twilson | 2010-10-13 15:45:15 -0700 (Wed, 13 Oct 2010)
| 21 lines Don't ignore frames that have been queued when
softhangup'd When an outgoing call is answered and hung up by the
far end *very* quickly, we may not read any frames and therefor
end up with a call that displays the wrong
disposition/DIALSTATUS. The reason is because ast_queue_hangup()
immediately sets the _softhangup flag on the channel and then
queues the HANGUP control frame, but __ast_read refuses to read
any frames if ast_check_hangup() indicates that a hangup request
has been made (which it will if _softhangup is set). So, we end
up losing control frames. This change makes __ast_read continue
to read frames even if a soft hangup has been requested. It
queues a hangup frame to make sure that __ast_read() will still
eventually return NULL. Much thanks to David Vossel for all of
the reviews, discussion, and help! (closes issue #16946) Reported
by: davidw Review: https://reviewboard.asterisk.org/r/740/
........
2010-10-13 15:29 +0000 [r291393] Russell Bryant <russell@digium.com>
* /, channels/chan_sip.c: Merged revisions 291392 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r291392 | russell | 2010-10-13 10:23:19 -0500 (Wed, 13 Oct 2010)
| 6 lines Lock pvt so pvt->owner can't disappear when queueing up
a frame. This fixes a crash due to a hangup race condition.
ABE-2601 ........
2010-10-12 17:20 +0000 [r291280] Leif Madsen <lmadsen@digium.com>
* configs/phoneprov.conf.sample: Add undocumented variables to
phoneprov.conf.sample (closes issue #18107) Reported by: lathama
Patches: phoneprov.conf.sample.diff uploaded by lathama (license
1028)
2010-10-12 17:05 +0000 [r291264] Tilghman Lesher <tlesher@digium.com>
* /, main/acl.c: Merged revisions 291263 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r291263 | tilghman | 2010-10-12 11:55:30 -0500 (Tue, 12 Oct 2010)
| 2 lines Oops, incorrect range (although unallocated at ARIN)
........
2010-10-12 16:07 +0000 [r291229] Leif Madsen <lmadsen@digium.com>
* configs/manager.conf.sample: Add documention that mentions
options are defined but not used. (Issue #18101)
2010-10-11 18:39 +0000 [r291073-291111] Richard Mudgett <rmudgett@digium.com>
* channels/chan_sip.c: Make exit from handle_request_do()
consistent.
* /, channels/chan_sip.c: Merged revisions 291109 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r291109 | rmudgett | 2010-10-11 13:29:43 -0500 (Mon, 11 Oct 2010)
| 1 line Add missing unlock to an exception condition in
reload_config(). ........
* main/cli.c: Fixed infinite loop in verbose/debug message output.
Setting the module/filename specific message level and then
changing it resulted in the linked list being looped on itself.
Traversing this linked list is an infinite loop if what you are
looking for is not in the list. Also plugged some CLI parsing
holes in the associated CLI command: * Removing a nonexistent
module from the list actually added it with a level of zero. *
Setting the non-module specific level to zero is now equivalent
to setting it to "off" as documented.
2010-10-08 02:45 +0000 [r290863] Jeff Peeler <jpeeler@digium.com>
* main/asterisk.c, /: Merged revisions 290862 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r290862 | jpeeler | 2010-10-07 21:35:29 -0500 (Thu, 07 Oct 2010)
| 9 lines Ensure editline cleanup occurs when Ctrl-C is pressed
at control console. A recent change was made to avoid a race
condition on shutdown which only called the end functions from
the console thread. However, when pressing Ctrl-C the quit
handler is called from the signal handler thread. (closes issue
#17698) Reported by: jmls ........
2010-10-07 20:57 +0000 [r290751] Jason Parker <jparker@digium.com>
* autoconf/ast_ext_lib.m4, /, configure,
include/asterisk/autoconfig.h.in: Merged revisions 290750 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r290750 | qwell | 2010-10-07 15:56:04 -0500 (Thu, 07 Oct 2010) |
9 lines Allow PRI to build properly when using --with-pri. Use
the directories found for the parent when using lib dependencies.
(closes issue #17314) Reported by: tzafrir Patches:
17314-withdeps.diff uploaded by qwell (license 4) ........
2010-10-07 10:53 +0000 [r290712] Russell Bryant <russell@digium.com>
* main/pbx.c: Don't crash when Set() is called without a value.
Review: https://reviewboard.asterisk.org/r/949/
2010-10-06 13:48 +0000 [r290396-290575] Tilghman Lesher <tlesher@digium.com>
* main/file.c: Allow streaming audio from a pipe. (closes issue
#18001) Reported by: jamicque Patches:
20100926__issue18001.diff.txt uploaded by tilghman (license 14)
Tested by: jamicque
* res/res_jabber.c, /: Merged revisions 290392 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r290392 | tilghman | 2010-10-05 15:20:07 -0500 (Tue, 05 Oct 2010)
| 8 lines Fix a crash by ensuring that we don't alter memory
after it's freed. (closes issue #17387) Reported by: jmls
Patches: 20100726__issue17387.diff.txt uploaded by tilghman
(license 14) Tested by: jmls ........
2010-10-05 19:54 +0000 [r290375] David Vossel <dvossel@digium.com>
* apps/app_directed_pickup.c: Fixes PickupChan() not working with
full channel name. (closes issue #18011) Reported by: schern
Patches: app_directed_pickup.c.2.patch uploaded by schern
(license 995) app_directed_pickup.c.trunk.patch uploaded by
schern (license 995) Tested by: schern, dvossel
2010-10-05 17:42 +0000 [r290324] Richard Mudgett <rmudgett@digium.com>
* contrib/valgrind.supp, /: Merged revisions 290323 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
................ r290323 | rmudgett | 2010-10-05 12:41:18 -0500
(Tue, 05 Oct 2010) | 11 lines Merged revision 258974 from
https://origsvn.digium.com/svn/asterisk/trunk .......... r258974
| diruggles | 2010-04-26 14:05:47 -0500 (Mon, 26 Apr 2010) | 4
lines Line 24 missed in compatibility fix in revision 233577
added a "fun:" prefix line 24 .......... ................
2010-10-04 23:14 +0000 [r290101-290254] Tilghman Lesher <tlesher@digium.com>
* pbx/ael/ael-test/ref.ael-test19,
pbx/ael/ael-test/ref.ael-vtest13, res/ael/pval.c, main/pbx.c,
pbx/ael/ael-test/ref.ael-vtest17,
pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3,
pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test5:
Change new pattern matcher to regard dashes the same as the old
pattern matcher -- as visual candy to be ignored. Also change the
AEL parser to not generate dashes within extensions, as those
dashes would be ignored. Update the AEL tests to match this
behavior. (closes issue #17366) Reported by: murf Patches:
20100727__issue17366.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
* /, configure, configure.ac: Merged revisions 290177 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r290177 | tilghman | 2010-10-04 15:15:26 -0500 (Mon, 04
Oct 2010) | 2 lines Fixing Mac OS X auto-builder. ........
* /, configure, configure.ac: Merged revisions 290100 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r290100 | tilghman | 2010-10-03 16:04:29 -0500 (Sun, 03
Oct 2010) | 2 lines Automatically re-run configure test for
menuselect, when the relevant makeopts settings change. ........
2010-10-02 08:52 +0000 [r289950] Olle Johansson <oej@edvina.net>
* main/manager.c, /: Merged revisions 289949 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r289949 | oej | 2010-10-02 10:50:05 +0200 (Lör, 02 Okt 2010) | 2
lines Add documentation for undocumented option to AMI action
originate ........
2010-10-02 04:45 +0000 [r289874] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c, /: Merged revisions 289873 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r289873 | tilghman | 2010-10-01 23:42:08 -0500 (Fri, 01
Oct 2010) | 8 lines When forwarding a message, a prepend means
that the filesystem will always have a better copy. (closes issue
#17803) Reported by: dpetersen Patches:
20100923__issue17803.diff.txt uploaded by tilghman (license 14)
Tested by: dpetersen ........
2010-10-01 23:01 +0000 [r289798] Jeff Peeler <jpeeler@digium.com>
* main/rtp.c, /, channels/chan_sip.c, include/asterisk/rtp.h:
Merged revisions 289797 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r289797 | jpeeler | 2010-10-01 17:58:38 -0500 (Fri, 01 Oct 2010)
| 15 lines Change RFC2833 DTMF event duration on end to report
actual elapsed time. The scenario here is with a non P2P early
media session. The reported time length of DTMF presses are
coming up short when sending to the remote side. Currently the
event duration is a running total that is incremented when
sending continuation packets. These continuation packets are only
triggered upon incoming media from the remote side, which means
that the running total probably is not going to end up matching
the actual length of time Asterisk received DTMF. This patch
changes the end event duration to be lengthened if it is detected
that the end event is going to come up short. Review:
https://reviewboard.asterisk.org/r/957/ ABE-2476 ........
2010-10-01 17:09 +0000 [r289704] Paul Belanger <paul.belanger@polybeacon.com>
* res/res_jabber.c, /, configs/jabber.conf.sample: Merged revisions
289703 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r289703 | pabelanger | 2010-10-01 13:03:11 -0400 (Fri, 01 Oct
2010) | 6 lines Disable debugging by default and reformat .config
file. Review: https://reviewboard.asterisk.org/r/929/ ........
2010-10-01 16:21 +0000 [r289700] Jeff Peeler <jpeeler@digium.com>
* /, channels/chan_sip.c: Merged revisions 289699 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r289699 | jpeeler | 2010-10-01 11:20:00 -0500 (Fri, 01 Oct 2010)
| 14 lines Ensure user portion of SIP URI matches dialplan when
using encoded characters. This commit takes a simliar approach to
288112 and checks the dialplan to determine the proper action for
an incoming contact header as to whether or not it should be
decoded or not. sip_new was blindly always decoding the
extension, which also caused the outgoing contact header to be
incorrect as well as failing to match the encoded extension in
the dialplan. (closes issue #17892) Reported by: wdoekes Patches:
bug17892-1.patch uploaded by jpeeler (license 325) Tested by:
wdoekes ........
2010-10-01 09:42 +0000 [r289622] schmitds <schmitds@localhost>:
* channels/chan_sip.c: don't iterate through all dialogs to find
and delete old subscribes On every incoming subscribe there is a
iteration through all dialogs to find old subscribes and delete
them. This is slow and not RFC conform. This was only needed in
1.2 cause a subscribe was not deleted when a dialog was
destroyed, after 1.4 a subscribe get removed when its dialog is
destroyed. (closes issue #17950) Reported by: schmidts Tested by:
schmidts Review: https://reviewboard.asterisk.org/r/901/
2010-09-30 19:51 +0000 [r289553] Matthew Nicholson <mnicholson@digium.com>
* channels/chan_sip.c: Properly handle channel allocation failures
duing invites with replaces. ABE-2588
2010-09-30 17:09 +0000 [r289501] Brett Bryant <bbryant@digium.com>
* /, res/res_agi.c: Merged revisions 289500 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r289500 | bbryant | 2010-09-30 13:08:20 -0400 (Thu, 30 Sep 2010)
| 11 lines res_agi.c:handle_getvariablefull() could recursively
lock a channel and not release it if an argument is the current
channel's name. (closes issue #17970) Reported by: mdu113
Patches: res_agi.c.diff3 uploaded by mdu113 (license 582) Tested
by: mdu113 Review: https://reviewboard.asterisk.org/r/947/
........
2010-09-30 15:37 +0000 [r289425] Russell Bryant <russell@digium.com>
* /, apps/app_sms.c: Merged revisions 289424 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r289424 | russell | 2010-09-30 10:34:29 -0500 (Thu, 30 Sep 2010)
| 8 lines Fix a crash in app_sms. Since the data being passed to
the generator callback is on the stack of the SMS() application,
we must ensure that the generator is stopped before the
application exits. ABE-2587 ........
2010-09-29 21:03 +0000 [r289339] Jason Parker <jparker@digium.com>
* main/channel.c, /, main/features.c: Merged revisions 289338 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r289338 | qwell | 2010-09-29 15:56:26 -0500 (Wed, 29 Sep 2010) |
8 lines Allow a manager originate to succeed on forwarded
devices. The timeout to wait for an answer was being set to 0
when a device forwarded to another extension. We don't always
need the timeout set like this, so make it an optional parameter,
and don't use it in this case. ABE-2544 ........
2010-09-29 20:24 +0000 [r289334] Leif Madsen <lmadsen@digium.com>
* configs/res_ldap.conf.sample: Update sample documentation to note
md5secret requirements.
2010-09-29 20:15 +0000 [r289332] Russell Bryant <russell@digium.com>
* res/res_config_ldap.c: Don't completely ignore md5secret from
LDAP if the value does not begin with {md5}. This fixes a problem
that lmadsen ran in to where md5secret was not working for him.
2010-09-29 15:04 +0000 [r289178] Matthew Nicholson <mnicholson@digium.com>
* main/channel.c, /: Merged revisions 289177 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r289177 | mnicholson | 2010-09-29 10:03:27 -0500 (Wed, 29 Sep
2010) | 8 lines Set the caller id on CDRs when it is set on the
parent channel. (closes issue #17569) Reported by: tbelder
Patches: 17569.diff uploaded by tbelder (license 618) Tested by:
tbelder ........
2010-09-28 18:14 +0000 [r289095] Brett Bryant <bbryant@digium.com>
* main/channel.c, /: Merged revisions 289094 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r289094 | bbryant | 2010-09-28 14:10:19 -0400 (Tue, 28 Sep 2010)
| 14 lines Fixes an issue with the Newchannel AMI event during
the Masquerading process. Fixes an issue with the Newchannel AMI
event during the Masquerading process, where no Newchannel AMI
event was generated for the psuedo channel used during the
masquerading process. (closes issue #17987) Reported by:
RadicAlish Patches: newchannel.patch.txt uploaded by RadicAlish
(license 1122) Tested by: RadicAlish Review:
https://reviewboard.asterisk.org/r/937/ ........
2010-09-24 15:37 +0000 [r288747] Terry Wilson <twilson@digium.com>
* channels/chan_local.c, /: Merged revisions 288746 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r288746 | twilson | 2010-09-24 08:26:09 -0700 (Fri, 24
Sep 2010) | 5 lines Don't fail a masquerade if it is already
being hung up This avoids noise on some Local channel situations
where we don't use /n. Thanks to Alec Davis for the suggestion.
........
2010-09-24 13:53 +0000 [r288637-288712] Tilghman Lesher <tlesher@digium.com>
* funcs/func_strings.c: Solaris won't printf a NULL. (closes issue
#18041) Reported by: asgaroth
* cdr/cdr_pgsql.c, /, configure, include/asterisk/autoconfig.h.in,
include/asterisk/compat.h, main/strcompat.c, configure.ac,
include/asterisk/channel.h: Merged revisions 288636 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r288636 | tilghman | 2010-09-23 22:20:24 -0500 (Thu, 23
Sep 2010) | 2 lines Solaris compatibility fixes ........
2010-09-22 23:10 +0000 [r288500] Terry Wilson <twilson@digium.com>
* channels/chan_local.c, /: Merged revisions 288499 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r288499 | twilson | 2010-09-22 16:00:30 -0700 (Wed, 22
Sep 2010) | 8 lines Don't let a Local channel get bridged to
itself If a local channel gets bridged to itself, it becomes
orphaned with no devices left to actually tell it to hang up.
This patch modifies local_fixup() to detect this case and deny
it. Review: https://reviewboard.asterisk.org/r/934 ........
2010-09-22 17:49 +0000 [r288344-288417] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 288416 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r288416 | dvossel | 2010-09-22 12:48:15 -0500 (Wed, 22 Sep 2010)
| 5 lines RFC3261 section 12.2 explicitly says out of order
requests are responded with a 500 Server Internal Error response.
ABE-2458 ........
* /, channels/chan_sip.c: Merged revisions 288343 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r288343 | dvossel | 2010-09-22 11:49:56 -0500 (Wed, 22 Sep 2010)
| 2 lines During check_pendings, if the dialog is terminated with
a CANCEL, change the invitestate to INV_CANCEL like in
sip_hangup. ........
2010-09-22 16:44 +0000 [r288340] Russell Bryant <russell@digium.com>
* main/asterisk.c, /: Merged revisions 288339 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r288339 | russell | 2010-09-22 11:39:16 -0500 (Wed, 22 Sep 2010)
| 11 lines Fix a 100% CPU consumption problem when setting
console=yes in asterisk.conf. The handling of -c and console=yes
should be the same, but they were not. When you specify -c, it
sets both a flag for console module and for asterisk not to
fork() off into the background. The handling of console=yes only
set console mode, so you would end up with a background process()
trying to run the Asterisk console and freaking out since it
didn't have anything to read input from. Thanks to beagles for
reporting and helping debug the problem! ........
2010-09-22 15:11 +0000 [r288267] Tilghman Lesher <tlesher@digium.com>
* cdr/cdr_pgsql.c, configs/cdr_pgsql.conf.sample, /, UPGRADE.txt:
Merged revisions 288265-288266 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r288265 | tilghman | 2010-09-22 09:48:04 -0500 (Wed, 22 Sep 2010)
| 9 lines Allow the encoding to be set, in case local charset
does not agree with database. (closes issue #16940) Reported by:
jamicque Patches: 20100827__issue16940.diff.txt uploaded by
tilghman (license 14) 20100921__issue16940__1.6.2.diff.txt
uploaded by tilghman (license 14) Tested by: jamicque ........
r288266 | tilghman | 2010-09-22 10:04:52 -0500 (Wed, 22 Sep 2010)
| 5 lines Document addition of encoding parameter. (issue #16940)
Reported by: jamicque ........
2010-09-22 00:03 +0000 [r288193] Richard Mudgett <rmudgett@digium.com>
* channels/chan_iax2.c, /: Merged revisions 288192 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r288192 | rmudgett | 2010-09-21 18:55:58 -0500 (Tue, 21
Sep 2010) | 26 lines In chan_iax2.c:schedule_delivery() calls
ast_bridged_channel() on an unlocked channel. Near the beginning
of schedule_delivery(), ast_bridged_channel() is called on
iaxs[fr->callno]->owner. However, the channel is not locked,
which can result in ast_bridged_channel() crashing should
owner->tech change to a technology that doesn't implement
bridged_channel. I also fixed the other calls to
ast_bridged_channel() in chan_iax2.c since the owner lock was not
held there either. Converted the existing channel deadlock
avoidance to use iax2_lock_owner(). Using the new function
simplified some awkward code. In the process of fixing the
locking on ast_bridged_channel(), I also found a memory leak in
socket_process() for v1.6.2 and v1.8. The local struct variable
ies.vars is not freed on early/abnormal function exits. (closes
issue #17919) Reported by: rain Patches: issue17919_v1.4.patch
uploaded by rmudgett (license 664) issue17919_w_leak_v1.6.2.patch
uploaded by rmudgett (license 664) issue17919_w_leak_v1.8.patch
uploaded by rmudgett (license 664) Review:
https://reviewboard.asterisk.org/r/926/ ........
2010-09-21 22:22 +0000 [r288147] Paul Belanger <paul.belanger@polybeacon.com>
* channels/chan_iax2.c: Setup timer before set_config(). (closes
issue #18019) Reported by: Netview Patches: issue_0018019.patch
uploaded by pabelanger (license 224) Tested by: Netview
2010-09-21 21:59 +0000 [r288113] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_sip.c: Merged revisions 288112 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r288112 | tilghman | 2010-09-21 16:58:13 -0500 (Tue, 21 Sep 2010)
| 15 lines Try both the encoded and unencoded subscription URI
for a match in hints. When a phone sends an encoded URI for a
subscription, the URI is not matched with the actual hint that is
in decoded format. For example, if we have an extension with a
hint that is named: "#5601" or "*5601", the subscription will
work fine if the phone subscribes with an already decoded URI,
but when it's decoded like "%255601" or "%2A5601", Asterisk is
unable to match it with the correct hint. (closes issue #17785)
Reported by: ramonpeek Patches: 20100831__issue17785.diff.txt
uploaded by tilghman (license 14) Tested by: ramonpeek ........
2010-09-21 19:46 +0000 [r288006] Brett Bryant <bbryant@digium.com>
* main/channel.c, /: Merged revisions 288005 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r288005 | bbryant | 2010-09-21 15:43:46 -0400 (Tue, 21 Sep 2010)
| 8 lines Add a check to fix a rare segmentation fault you'd get
if ast_frdup couldn't allocate memory on the first frame being
queued in ast_queue_frame. (closes issue #17882) Reported by:
seanbright Tested by: seanbright ........
2010-09-21 19:07 +0000 [r287934] Tilghman Lesher <tlesher@digium.com>
* main/asterisk.c, /: Merged revisions 287933 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r287933 | tilghman | 2010-09-21 14:07:07 -0500 (Tue, 21 Sep 2010)
| 2 lines Less than zero is an error, not any non-zero value.
........
2010-09-20 23:58 +0000 [r287759] Brett Bryant <bbryant@digium.com>
* /, apps/app_meetme.c: Merged revisions 287758 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r287758 | bbryant | 2010-09-20 19:57:08 -0400 (Mon, 20 Sep 2010)
| 16 lines Fix misvalidation of meetme pins in conjunction with
the 'a' MeetMe flag. When using the 'a' MeetMe flag and having a
user and admin pin setup for your conference, using the user pin
would gain you admin priviledges. Also, when no user pin was set,
an admin pin was, the 'a' MeetMe flag wasn't used, and the user
tried to enter a conference then they were still prompted for a
pin and forced to hit #. (closes issue #17908) Reported by: kuj
Patches: pins_2.patch uploaded by kuj (license 1111) Tested by:
kuj Review: [full review board URL with trailing slash] ........
2010-09-20 23:16 +0000 [r287685] Alec L Davis <sivad.a@paradise.net.nz>
* main/channel.c: ast_channel_masquerade: Avoid recursive
masquerades. Check all 4 combinations of (original/clonechan) *
(masq/masqr). Initially original->masq and clonechan->masqr were
only checked. It's possible with multiple masq's planned - and
not yet executed, that the 'original' chan could already have
another masq'd into it - thus original->masqr would be set, that
masqr would lost. Likewise for the clonechan->masq. (closes issue
#16057;#17363) Reported by: amorsen;davidw,alecdavis Patches:
based on bug16057.diff4.txt uploaded by alecdavis (license 585)
Tested by: ramonpeek, davidw, alecdavis
2010-09-20 21:28 +0000 [r287642] Jason Parker <jparker@digium.com>
* channels/chan_skinny.c: Don't crash when parking a non-bridged
call. (closes issue #17680) Reported by: jmhunter Patches:
chan_skinny-park-v1.txt uploaded by DEA (license 3) Tested by:
jmhunter, DEA
2010-11-02 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.14 Released.
2010-09-20 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.14-rc1 Released.
2010-09-20 15:56 +0000 [r287556-287558] Matthew Nicholson <mnicholson@digium.com>
* main/pbx.c, /: Use ast_str when processing hint state changes
Merged revisions 287555 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r287555 | mnicholson | 2010-09-20 10:48:14 -0500 (Mon, 20 Sep
2010) | 5 lines Use ast_dynamic_str when processing hint state
changes (related to issue #17928) Reported by: mdu113 ........
* /: Revert r287556.
* /: Use ast_str when processing hint state changes Merged
revisions 287555 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r287555 | mnicholson | 2010-09-20 10:48:14 -0500 (Mon, 20 Sep
2010) | 5 lines Use ast_dynamic_str when processing hint state
changes (related to issue #17928) Reported by: mdu113 ........
2010-09-19 16:06 +0000 [r287470] Olle Johansson <oej@edvina.net>
* main/manager.c, /: Merged revisions 287469 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r287469 | oej | 2010-09-19 17:56:50 +0200 (Sön, 19 Sep 2010) | 7
lines Make sure we always free variables properly in manager
originate. (closes issue #17891) reported, solved and tested by
oej Review: https://reviewboard.asterisk.org/r/869/ ........
2010-09-17 21:08 +0000 [r287387] Tilghman Lesher <tlesher@digium.com>
* apps/app_queue.c, /: Merged revisions 287386 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r287386 | tilghman | 2010-09-17 16:06:03 -0500 (Fri, 17 Sep 2010)
| 7 lines Blank columns should get set on reload, not ignored.
(closes issue #16893) Reported by: haakon Patches:
20100818__issue16893.diff.txt uploaded by tilghman (license 14)
........
2010-09-17 13:36 +0000 [r287308] Matthew Nicholson <mnicholson@digium.com>
* main/pbx.c, /: Merged revisions 287307 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r287307 | mnicholson | 2010-09-17 08:34:34 -0500 (Fri, 17 Sep
2010) | 5 lines Use ast_strdup() instead of ast_strdupa() while
processing in ast_hint_state_changed(). (related to issue #17928)
Reported by: mdu113 ........
2010-09-16 22:12 +0000 [r287198] Jason Parker <jparker@digium.com>
* contrib/init.d/rc.debian.asterisk, /: Merged revisions 287197 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r287197 | qwell | 2010-09-16 17:12:30 -0500 (Thu, 16 Sep 2010) |
7 lines Add LSB headers for Debian init script, since Debian will
complain if it isn't there. Headers were taken from trunk.
(closes issue #17958) Reported by: javyer ........
2010-09-16 20:06 +0000 [r287115-287119] Matthew Nicholson <mnicholson@digium.com>
* main/pbx.c, /: Merged revisions 287118 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r287118 | mnicholson | 2010-09-16 15:04:46 -0500 (Thu, 16 Sep
2010) | 8 lines Don't limit hint processing in
ast_hint_state_changed() to AST_MAX_EXTENSION length strings.
(closes issue #17928) Reported by: mdu113 Patches:
20100831__issue17928.diff.txt uploaded by tilghman (license 14)
Tested by: mdu113 ........
* main/cdr.c, /: Merged revisions 287114 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r287114 | mnicholson | 2010-09-16 14:52:39 -0500 (Thu, 16 Sep
2010) | 8 lines Don't stop printing cdr variables if we encounter
one with a blank name or value. (closes issue #17900) Reported
by: under Patches: core-show-channel-cdr-fix1.diff uploaded by
mnicholson (license 96) Tested by: mnicholson ........
2010-09-15 20:28 +0000 [r286998] Jeff Peeler <jpeeler@digium.com>
* apps/app_voicemail.c, /: Merged revisions 286941 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r286941 | jpeeler | 2010-09-15 15:08:52 -0500 (Wed, 15
Sep 2010) | 7 lines Ensure mailbox is not filled to capacity
before doing message forwarding. Specifically, before prompting
to record a prepended message the capacity is checked first. If
the mailbox is full the extension will be reprompted. ABE-2517
........
2010-09-14 19:27 +0000 [r286681-286757] Matthew Nicholson <mnicholson@digium.com>
* /, channels/chan_sip.c: Merged revisions 286756 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r286756 | mnicholson | 2010-09-14 14:26:18 -0500 (Tue, 14 Sep
2010) | 13 lines Don't clear the username from a realtime
database when a registration expires. Non-realtime chan_sip does
not clear the username from memory when a registration expiries
so realtime probably shouldn't either. (closes issue #17551)
Reported by: ricardolandim Patches:
reg-expiry-username-1.4-fix1.diff uploaded by mnicholson (license
96) reg-expiry-username-1.6.2-fix1.diff uploaded by mnicholson
(license 96) reg-expiry-username-1.8-fix1.diff uploaded by
mnicholson (license 96) reg-expiry-username-trunk-fix1.diff
uploaded by mnicholson (license 96) Tested by: ricardolandim,
mnicholson ........
* main/channel.c, /: Merged revisions 286679 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r286679 | mnicholson | 2010-09-14 13:00:01 -0500 (Tue, 14 Sep
2010) | 7 lines Only drop duplicate answer frames if the channel
is bridged. Back in r3710 ast_read() was modified to drop answer
frames on channels that were in the UP state. This modification
prevented bridges that were up before the answer from being
broken and reestablished by an ANSWER control frame. That change
also prevents pickup of channels called from the ast_dial
framework from working properly. The ast_dial framework expects
to see an ANSWER frame after dialing and the pickup code queues
one but ast_read() drops it. This new change only drops ANSWER
frames when the channel is bridged, allowing the answer queued by
the pickup code to properly pass through ast_read() on to the
ast_dial framework. ABE-2473 (related to issue #2342) ........
2010-09-14 05:06 +0000 [r286527-286587] Tilghman Lesher <tlesher@digium.com>
* contrib/realtime/mysql/voicemail_messages.sql (added),
contrib/realtime/mysql/voicemail_data.sql (added): Add
documentation on missing backend tables for Voicemail
* main/features.c: C precedence got me
* main/features.c: Refactor conversion to ast_poll() to fix
callparking regression.
2010-09-13 19:38 +0000 [r286456] Jason Parker <jparker@digium.com>
* channels/chan_sip.c: Remove "Internal IP" from sip show settings,
as it's not at all useful to display. (closes issue #17840)
Reported by: oej
2010-09-11 17:05 +0000 [r286268] Olle Johansson <oej@edvina.net>
* /, main/file.c: Merged revisions 286267 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r286267 | oej | 2010-09-11 18:59:20 +0200 (Lör, 11 Sep 2010) | 4
lines Handle error response when we can't make file compatible
Review: https://reviewboard.asterisk.org/r/911/ ........
2010-09-10 22:56 +0000 [r286223] Terry Wilson <twilson@digium.com>
* channels/chan_local.c, /: Merged revisions 286222 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r286222 | twilson | 2010-09-10 17:54:23 -0500 (Fri, 10
Sep 2010) | 1 line Return -1 if chan_local doesn't support an
option ........
2010-09-10 20:55 +0000 [r286117] Paul Belanger <paul.belanger@polybeacon.com>
* channels/chan_iax2.c, /: Merged revisions 286114 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r286114 | pabelanger | 2010-09-10 16:35:08 -0400 (Fri,
10 Sep 2010) | 4 lines Load iax.conf before registering any
functions/applications/actions. Review:
https://reviewboard.asterisk.org/r/914/ ........
2010-09-10 20:42 +0000 [r286116] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 286113 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r286113 | rmudgett | 2010-09-10 15:33:16 -0500 (Fri, 10
Sep 2010) | 11 lines An outgoing call may not get hung up if a
pre-connect incoming ISDN call is disconnected. If the ISDN link
a pre-connect incoming call is using fails or is reset, the
outgoing leg may not hang up or be delayed in hanging up.
(Causes: PRI_CAUSE_NETWORK_OUT_OF_ORDER,
PRI_CAUSE_DESTINATION_OUT_OF_ORDER, and
PRI_CAUSE_NORMAL_TEMPORARY_FAILURE.) Just hang up the call if the
incoming call leg hangs up before connecting for any reason. It
makes no sense to send a BUSY or CONGESTION control frame to the
outgoing call leg under these circumstances. ........
2010-09-10 20:35 +0000 [r286115] Terry Wilson <twilson@digium.com>
* include/asterisk/pbx.h, include/asterisk/frame.h,
channels/chan_local.c, /, funcs/func_channel.c,
include/asterisk/channel.h: Merged revisions 286059 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r286059 | twilson | 2010-09-10 14:25:08 -0500 (Fri, 10
Sep 2010) | 16 lines Inherit CHANNEL() writes to both sides of a
Local channel Having Local (/n) channels as queue members and
setting the language in the extension with
Set(CHANNEL(language)=fr) sets the language on the Local/...,2
channel. Hold time report playbacks happen on the Local/...,1
channel and therefor do not play in the specified language. This
patch modifies func_channel_write to call the setoption callback
and pass the CHANNEL() write info to the callback. chan_local
uses this information to look up the other side of the channel
and apply the same changes to it. (closes issue #17673) Reported
by: Guggemand Review: https://reviewboard.asterisk.org/r/903/
........
2010-09-10 18:30 +0000 [r285930-286024] Tilghman Lesher <tlesher@digium.com>
* tests/test_heap.c, /, main/test.c: Merged revisions 286023 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r286023 | tilghman | 2010-09-10 13:22:04 -0500 (Fri, 10 Sep 2010)
| 2 lines Missing newline ........
* include/asterisk/select.h: Another fix for Mac OS X. While trying
to fix this the "right" way, I wandered into dependency hell. Two
hours later, I backed out, and just removed the offending code.
ast_inline_api only goes one level deep and then it breaks. Ouch.
* tests/test_poll.c, include/asterisk/select.h, /, configure,
include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
285889 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r285889 | tilghman | 2010-09-09 19:13:45 -0500 (Thu, 09 Sep 2010)
| 7 lines Fix Mac OS X build. This also fixes a rather grievous
calculation error for the offset of ast_fdset, which was masked
on Linux and FreeBSD, because these platforms check the first 256
FDs regardless of the bitmask setting (due to backwards
compatibility). ........
2010-09-09 22:49 +0000 [r285818] Paul Belanger <paul.belanger@polybeacon.com>
* /, codecs/gsm/Makefile: Merged revisions 285817 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r285817 | pabelanger | 2010-09-09 18:34:35 -0400 (Thu, 09 Sep
2010) | 8 lines GCC 4.2.x optimizations result in improper
behavior of GSM codec (closes issue #17688) Reported by:
pprindeville Patches: asterisk-trunk-bugid11243.patch uploaded by
pprindeville (license 347) Tested by: mkeuter, pprindeville
........
2010-09-09 20:09 +0000 [r285744] Jason Parker <jparker@digium.com>
* main/channel.c, /: Merged revisions 285742 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r285742 | qwell | 2010-09-09 15:06:31 -0500 (Thu, 09 Sep 2010) |
9 lines Transmit silence when reading DTMF in ast_readstring.
Otherwise, you could get issues with DTMF timeouts causing
hangups. (closes issue #17370) Reported by: makoto Patches:
channel-readstring-silence-generator.patch uploaded by makoto
(license 38) ........
2010-09-09 18:50 +0000 [r285639-285710] Brett Bryant <bbryant@digium.com>
* main/pbx.c: Fixes an issue with dialplan pattern matching where
the specificity for pattern ranges and pattern special characters
was inconsistent. (closes issue #16903) Reported by: Nick_Lewis
Patches: pbx.c-specificity.patch uploaded by Nick Lewis (license
657) Tested by: Nick_Lewis
* res/res_musiconhold.c, /: Merged revisions 285638 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r285638 | bbryant | 2010-09-09 13:20:17 -0400 (Thu, 09
Sep 2010) | 7 lines Fixes an issue with MOH where it doesn't
recover cleanly when it can't play a file and would just stop,
instead of continuing to find the next playable file in the MOH
class. (closes issue #17807) Reported by: kshumard Review:
https://reviewboard.asterisk.org/r/910/ ........
2010-09-08 22:11 +0000 [r285563-285567] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 285566 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r285566 | dvossel | 2010-09-08 17:07:31 -0500 (Wed, 08 Sep 2010)
| 2 lines In retrans_pkt, do not unlock pvt until the end of the
function on a transmit failure. ........
* channels/chan_sip.c: Fixes interoperability problems with session
timer behavior in Asterisk. CHANGES: 1. Never put "timer" in
"Require" header. This is not to our benefit and RFC 4028 section
7.1 even warns against it. It is possible for one endpoint to
perform session-timer refreshes while the other endpoint does not
support them. If in this case the end point performing the
refreshing puts "timer" in the Require field during a refresh,
the dialog will likely get terminated by the other end. 2. Change
the behavior of 'session-timer=accept' in sip.conf (which is the
default behavior of Asterisk with no session timer configuration
specified) to only run session-timers as result of an incoming
INVITE request if the INVITE contains an "Session-Expires"
header... Asterisk is currently treating having the "timer"
option in the "Supported" header as a request for session timers
by the UAC. I do not agree with this. Session timers should only
be negotiated in "accept" mode when the incoming INVITE supplies
a "Session-Expires" header, otherwise RFC 4028 says we should
treat a request containing no "Session-Expires" header as a
session with no expiration. Below I have outlined some situations
and what Asterisk's behavior is. The table reflects the behavior
changes implemented by this patch. SITUATIONS: -Asterisk as UAS
1. Incoming INVITE: NO "Session-Expires" 2. Incoming INVITE: HAS
"Session-Expires" -Asterisk as UAC 3. Outgoing INVITE: NO
"Session-Expires". 200 Ok Response HAS "Session-Expires" header
4. Outgoing INVITE: NO "Session-Expires". 200 Ok Response NO
"Session-Expires" header 5. Outgoing INVITE: HAS
"Session-Expires". Active - Asterisk will have an active refresh
timer regardless if the other endpoint does. Inactive - Asterisk
does not have an active refresh timer regardless if the other
endpoint does. XXXXXXX - Not possible for mode.
______________________________________ |SITUATIONS |
'session-timer' MODES | |___________|________________________| |
| originate | accept | |-----------|------------|-----------| |1.
| Active | Inactive | |2. | Active | Active | |3. | XXXXXXXX |
Active | |4. | XXXXXXXX | Inactive | |5. | Active | XXXXXXXX |
-------------------------------------- (closes issue #17005)
Reported by: alexrecarey
2010-09-08 20:56 +0000 [r285532] Brett Bryant <bbryant@digium.com>
* apps/app_meetme.c: Fixes a bug with MeetMe where after announcing
the amount of time left in a conference, if music on hold was
playing, it doesn't restart. (closes issue #17408) Reported by:
sysreq Patches: asterisk-issue-17408_fixed.patch uploaded by
sysreq (license 1009) Tested by: sysreq
2010-09-08 20:42 +0000 [r285526-285529] Jason Parker <jparker@digium.com>
* res/res_musiconhold.c, include/asterisk/astobj2.h: Follow coding
guidelines in moh rescan fix. Also fix the documentation that got
me in trouble.
* res/res_musiconhold.c: Fixes issue where moh files were no longer
rescanned during a reload. (closes issue #16744) Reported by: pj
Patches: 16744-reload.diff uploaded by qwell (license 4) Tested
by: qwell
2010-09-07 20:31 +0000 [r285267-285366] Tilghman Lesher <tlesher@digium.com>
* pbx/pbx_config.c, /: Merged revisions 285365 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r285365 | tilghman | 2010-09-07 15:30:22 -0500 (Tue, 07 Sep 2010)
| 9 lines Catch invalid extensions at the parser, instead of
making the core deal with them. (closes issue #17794) Reported
by: PavelL Patches: 20100820__issue17794__1.6.2.diff.txt uploaded
by tilghman (license 14) 20100820__issue17794__1.4.diff.txt
uploaded by tilghman (license 14) Tested by: PavelL ........
* main/poll.c, /: Merged revisions 285266 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r285266 | tilghman | 2010-09-07 14:04:50 -0500 (Tue, 07 Sep 2010)
| 4 lines Use poll, if indicated to do so, in the ast_poll2
implementation. This fixes the unit tests on FreeBSD 8.0.
........
2010-09-07 17:49 +0000 [r285196] Brett Bryant <bbryant@digium.com>
* apps/app_voicemail.c, /: Merged revisions 285194 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r285194 | bbryant | 2010-09-07 13:45:41 -0400 (Tue, 07
Sep 2010) | 10 lines Fixes voicemail.conf issues where mailboxes
with passwords that don't precede a comma would throw unnecessary
error messages. (closes issue #15726) Reported by: 298 Patches:
M15726.diff uploaded by junky (license 177) Tested by: junky
Review: [full review board URL with trailing slash] ........
2010-09-06 06:55 +0000 [r285089] Tilghman Lesher <tlesher@digium.com>
* makeopts.in, /, BSDmakefile (added): Merged revisions 285088 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r285088 | tilghman | 2010-09-06 01:54:18 -0500 (Mon, 06 Sep 2010)
| 2 lines Silly convenience script for BSD platforms. ........
2010-09-03 18:15 +0000 [r284958] Brett Bryant <bbryant@digium.com>
* channels/chan_iax2.c: This is a patch provided for issue #17935
to add the ActionID to the IAXregistry AMI response. (closes
issue #17935) Reported by: alexkuklin Patches: iaxshowreg
uploaded by alexkuklin (license 1115) Tested by: alexkuklin
2010-09-03 16:20 +0000 [r284897] Terry Wilson <twilson@digium.com>
* apps/app_chanspy.c, /: Merged revisions 284881 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r284881 | twilson | 2010-09-03 11:10:23 -0500 (Fri, 03 Sep 2010)
| 5 lines Properly detect when a sound file doesn't exist
ast_fileexists returns -1 for error and 0 for a non-existant
file. The existing code treated missing files as though they
existed. ........
2010-09-02 20:54 +0000 [r284778] Brett Bryant <bbryant@digium.com>
* main/manager.c, /: Merged revisions 284777 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r284777 | bbryant | 2010-09-02 16:25:03 -0400 (Thu, 02 Sep 2010)
| 7 lines Fixes a bug in manager.c where the default
configuration values weren't reset when the manager configuration
was reloaded. (closes issue #17917) Reported by: lmadsen Review:
https://reviewboard.asterisk.org/r/883/ ........
2010-09-02 16:48 +0000 [r284704] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 284703 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r284703 | dvossel | 2010-09-02 11:47:15 -0500 (Thu, 02 Sep 2010)
| 7 lines Removed relatedpeer code from sip_autodestruct Handling
of the relatedpeer structure associated with a sip_pvt should be
done during the final sip_destruction function, not in
sip_autodestruct. ........
2010-09-02 16:07 +0000 [r284399-284665] Tilghman Lesher <tlesher@digium.com>
* channels/chan_usbradio.c: Fixing build.
* apps/app_queue.c: Don't reset queue stats on a module reload.
(closes issue #17535) Reported by: raarts Patches:
20100819__issue17535.diff.txt uploaded by tilghman (license 14)
* configure, include/asterisk/autoconfig.h.in: Failed to rerun
bootstrap.sh after last commit
* res/res_jabber.c, main/rtp.c, main/poll.c,
include/asterisk/select.h (added), channels/chan_usbradio.c,
channels/chan_phone.c, channels/chan_misdn.c, main/features.c,
include/asterisk/poll-compat.h, tests/test_poll.c (added),
main/asterisk.c, utils/clicompat.c, res/res_ais.c, /,
configure.ac, channels/console_video.c,
include/asterisk/channel.h: Merged revisions 284478 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r284478 | tilghman | 2010-09-01 13:49:11 -0500 (Wed, 01
Sep 2010) | 11 lines Ensure that all areas that previously used
select(2) now use poll(2), with implementations that need poll(2)
implemented with select(2) safe against 1024-bit overflows. This
is a followup to the fix for the pthread timer in 1.6.2 and
beyond, fixing a potential crash bug in all supported releases.
(closes issue #17678) Reported by: russell Branch:
https://origsvn.digium.com/svn/asterisk/team/tilghman/ast_select
Review: https://reviewboard.asterisk.org/r/824/ ........
* res/res_config_pgsql.c: Don't warn on floats and timestamps
(closes issue #17082) Reported by: coolmig
* /, channels/chan_sip.c: Merged revisions 284393 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r284393 | tilghman | 2010-08-31 15:13:21 -0500 (Tue, 31 Aug 2010)
| 7 lines Don't send a devstate change on poke_noanswer if the
state did not change. (closes issue #17741) Reported by: schmidts
Patches: chan_sip.c.patch uploaded by schmidts (license 1077)
........
2010-08-31 18:59 +0000 [r284317] Leif Madsen <lmadsen@digium.com>
* configs/say.conf.sample, /: Merged revisions 284316 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r284316 | lmadsen | 2010-08-31 13:57:59 -0500 (Tue, 31
Aug 2010) | 7 lines Update say.conf.sample to match the rules in
say.c (closes issue #17835) Reported by: RoadKill Patches:
say.conf.sample.patch.rules uploaded by RoadKill (license 933)
Tested by: RoadKill ........
2010-08-30 22:27 +0000 [r284280] Tilghman Lesher <tlesher@digium.com>
* apps/app_festival.c: Fix 3 coding errors: 1) After we close FD,
we should not be trying to write to it. 2) Call _exit(0), not
exit(0), to avoid running shutdown routines in a child. 3) Use
endian, not processor, detection to ensure bytes are written in
the correct order. (closes issue #15706) Reported by: modelnine
Patches: asterisk-1.6.1.1-festival-debug.patch uploaded by
modelnine (license 865) Tested by: gmartinez
2010-08-27 22:27 +0000 [r284002] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 283960 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r283960 | dvossel | 2010-08-27 17:17:26 -0500 (Fri, 27 Aug 2010)
| 8 lines Parse all "Accept" headers for SIP SUBSCRIBE requests.
(closes issue #17758) Reported by: ibc Patches:
multiple_accept_headers_1.4.diff uploaded by dvossel (license
671) ........
2010-08-27 20:30 +0000 [r283881] Jason Parker <jparker@digium.com>
* res/res_config_pgsql.c, res/res_config_odbc.c, /: Merged
revisions 283880 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r283880 | qwell | 2010-08-27 15:29:11 -0500 (Fri, 27 Aug 2010) |
8 lines Fix issue with decoding ^-escaped characters in realtime.
(closes issue #17790) Reported by: denzs Patches:
17790-chunky.diff uploaded by qwell (license 4) Tested by: qwell,
denzs ........
2010-08-26 15:24 +0000 [r283381-283691] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 283690 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r283690 | dvossel | 2010-08-26 10:22:28 -0500 (Thu, 26 Aug 2010)
| 19 lines Fixed how Asterisk destroys a dialog on channel hangup
before invite receives a response. If an ast_channel with a SIP
tech pvt hangs up before the sip dialog gets a response to its
outgoing INVITE, Asterisk used to pretend_ack the INVITE. This is
not rfc compliant and results in confusion at the other endpoint.
sip_pretend_ack will ack and remove all the packets in the
retransmit queue. This means that the INVITE will stop
retransmitting, and that any response to that INVITE that comes
after the pretend_ack occurs will be ignored. Instead of faking
any sort of acknowledgement for an outgoing INVITE during an
internal hangup, we should let the protocol stack process the
INVITE transaction and terminate the dialog properly. This is
achieved by setting the PENDING_BYE flag. When this flag is used,
once the dialog proceeds to an escapable state the transaction
will either be canceled with a SIP_CANCEL or completed followed
immediately by a BYE. Attempting to do this any other way is
incorrect. If the endpoint is not responding to the INVITE
request, the INVITE must continue to be retransmitted until it
times out which will result in the dialog being destroyed.
........
* channels/chan_sip.c: Add to and from tags to NOTIFY dialog-info
xml body so pickup can occur. When pedantic mode is used, the
dialog-info xml generated during a ringing event must contain the
to and from tag values. Otherwise if a pickup occurs using INVITE
with replaces, Astrisk will not be able to locate the
subscription.
* channels/chan_sip.c: Asterisk will not advertise session timers
are supported when 'session-timers=refuse' is used. Asterisk now
dynamically builds the "Supported" header depending on what is
enabled/disabled in sip.conf. Session timers used to always be
advertised as being supported even when they were disabled in the
configuration. This caused problems with some end points. (issue
#17005)
* /, channels/chan_sip.c: Merged revisions 283380 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r283380 | dvossel | 2010-08-24 11:01:51 -0500 (Tue, 24 Aug 2010)
| 11 lines This fix makes sure the ast_channel hangs up correctly
when the dialog's PENDING_BYE flag is set. When the pending bye
flag is used, it is possible that the dialog will terminate and
leave the sip_pvt->owner channel up. This is because we never
hangup the ast_channel after sending the SIP_BYE request. When we
receive the response for the SIP_BYE we set need_destroy which we
would expect to destroy the dialog on the next do_monitor loop,
but this is not the case. The dialog will only be destroyed once
the owner is hungup even with the need_destroy flag set. This
patch sets the softhangup flag on the ast_channel when a SIP_BYE
request is sent as a result of the pending bye flag. ........
2010-08-23 21:32 +0000 [r283318] Tilghman Lesher <tlesher@digium.com>
* cdr/cdr_odbc.c, cdr/cdr_adaptive_odbc.c: CDR drivers depend upon
res_odbc, not directly on the ODBC libraries
2010-09-15 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.13 released.
* Incorrect .version and ChangeLog files updated. Re-release
of Asterisk 1.6.2.12 with corrections and version
number bump.
2010-09-15 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.12 released.
2010-08-23 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.12-rc1 Released.
2010-08-20 16:48 +0000 [r283049-283124] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 283123 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
................ r283123 | rmudgett | 2010-08-20 11:46:22 -0500
(Fri, 20 Aug 2010) | 9 lines Merged revision 278274 from
https://origsvn.digium.com/svn/asterisk/trunk .......... r278274
| rmudgett | 2010-07-20 17:38:13 -0500 (Tue, 20 Jul 2010) | 1
line Reference correct struct member for unlikely event
PRI_EVENT_CONFIG_ERR. .......... ................
* channels/chan_dahdi.c, /: Merged revisions 283048 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r283048 | rmudgett | 2010-08-20 10:24:36 -0500 (Fri, 20
Aug 2010) | 22 lines Q931 - Sending PROGRESS after sending
ALERTING is a protocol error The PRI layer in chan_dadhi will
check if a PROGRESS message has already been sent, and not allow
sending another (although that is technically allowed by the Q931
spec), however it does not protect against sending an ALERTING
and then sending a PROGRESS message, which is a violation of the
specification. Most switches don't seem to care too deeply about
this, but some do, and will disconnect the call when receiving
this invalid sequence. Protocol specification reference:
T-REC-Q.931-199805-I page 223, "Figure A.5/Q.931 -- Overview
protocol control (network side) point-point (sheet 3 of 8)"
(closes issue #17874) Reported by: nic_bellamy Patches:
asterisk-1.4-r282537_no-progress-after-alerting.patch uploaded by
nic bellamy (license 299)
asterisk-1.6.2-r282537_no-progress-after-alerting.patch uploaded
by nic bellamy (license 299)
asterisk-trunk-r282537_no-progress-after-alerting.patch uploaded
by nic bellamy (license 299) ........
2010-08-19 21:05 +0000 [r282890-282894] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 282893 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r282893 | dvossel | 2010-08-19 16:03:24 -0500 (Thu, 19 Aug 2010)
| 11 lines tos_sip option was not being set correctly When
tos_sip is used, the tos of the sip socket is only set correctly
if the socket binding changes on a reload. If the binding stays
the same but the TOS changes, the new tos value would not take
into effect. This patch fixes that. (closes issue #17712)
Reported by: nickb ........
* channels/chan_sip.c: fixes sip peer memory leaks in the
peer_by_ip table (issue #17798)
2010-08-19 19:44 +0000 [r282859] Matthew Nicholson <mnicholson@digium.com>
* channels/chan_sip.c: Merged revisions 277944 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r277944 | pabelanger | 2010-07-19 15:56:07 -0500 (Mon, 19 Jul
2010) | 16 lines Regression with T.38 negotiation Prior to
1.4.26.3 T.38 negotiation worked properly, in the case of the
reporter. (issue #16852) Reported by: cfc (closes issue #16705)
Reported by: mpiazzatnetbug Patches: issue16705_2.diff uploaded
by ebroad (license 878) Tested by: vrban, ebroad, c0rnoTa,
samdell3 Review: https://reviewboard.asterisk.org/r/754/ ........
2010-08-19 02:14 +0000 [r282730] Terry Wilson <twilson@digium.com>
* configs/sip.conf.sample, /: Merged revisions 282729 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r282729 | twilson | 2010-08-18 21:12:55 -0500 (Wed, 18
Aug 2010) | 2 lines Add some documentation about codec
negotiation to sip.conf ........
2010-08-18 14:28 +0000 [r282668] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: fixes crash with notifycid (closes issue
#17868) Reported by: francesco_r Patches: issue_17868.diff
uploaded by dvossel (license 671) Tested by: francesco_r
2010-08-18 07:43 +0000 [r282607] Tilghman Lesher <tlesher@digium.com>
* channels/chan_dahdi.c: Don't warn on callerid when completely
text, instead of numeric with localdialplan prefixes. (closes
issue #16770) Reported by: jamicque Patches:
20100413__issue16770.diff.txt uploaded by tilghman (license 14)
20100811__issue16770.diff.txt uploaded by tilghman (license 14)
Tested by: jamicque
2010-08-17 21:35 +0000 [r282576] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: fixes no default transport for temp peer
creation in chan_sip (closes issue #17829) Reported by: falves11
Patches: issue_17829.rev1.txt uploaded by russell (license 2)
issue_17829.diff uploaded by dvossel (license 671) Tested by:
falves11
2010-08-16 18:00 +0000 [r282469] Leif Madsen <lmadsen@digium.com>
* doc/tex/asterisk.tex, doc/tex/sounds.tex (added): Add information
about creating sounds files using the sounds tools publically
available so that others can create their own sounds prompts
using the same tools we use to generate sounds releases. This
allows people creating their own prompts to sound consistent with
the prompts available from the open source project. SWP-595
2010-08-16 17:32 +0000 [r282467] Terry Wilson <twilson@digium.com>
* main/channel.c, /: Merged revisions 282430 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r282430 | twilson | 2010-08-16 12:06:37 -0500 (Mon, 16 Aug 2010)
| 16 lines Send a SRCCHANGE indication when we masquerade
Masquerading a channel means that the src of the audio is
potentially changing, so send a SRCCHANGE so that RTP-based media
streams can get a new SSRC generated to reflect the change.
Original patch by addix (along with lots of testing--thanks!).
(closes issue #17007) Reported by: addix Patches:
1001-reset-SSRC-original-channel.diff uploaded by addix (license
1006) srcchange.diff uploaded by twilson (license 396) Tested by:
addix, twilson Review: https://reviewboard.asterisk.org/r/862/
........
2010-08-13 18:54 +0000 [r282235] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: only do magic pickup when notifycid is
enabled A new way of doing BLF pickup was introduced into 1.6.2.
This feature adds a call-id value into the XML of a SIP_NOTIFY
message sent to alert a subscriber that a device is ringing. This
option should only be enabled when the new 'notifycid' option is
set... but this was not the case. Instead the call-id value was
included for every RINGING Notify message, which caused a
regression for people who used other methods for call pickup.
(closes issue #17633) Reported by: urosh Patches: chan_sip.txt
uploaded by urosh (license ) blf_cid_issue.diff uploaded by
dvossel (license 671) Tested by: dvossel, urosh, okrief,
alecdavis
2010-08-12 22:50 +0000 [r282130] Jason Parker <jparker@digium.com>
* pbx/pbx_config.c, /: Merged revisions 282129 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r282129 | qwell | 2010-08-12 17:49:28 -0500 (Thu, 12 Aug 2010) |
1 line Register CLI commands before parsing config, in case there
is a config error. ........
2010-08-12 03:01 +0000 [r281912] Jeff Peeler <jpeeler@digium.com>
* main/channel.c, /: Merged revisions 281911 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r281911 | jpeeler | 2010-08-11 22:00:14 -0500 (Wed, 11 Aug 2010)
| 20 lines Ensure SSRC is changed when media source is changed to
resolve audio delay. This change causes the SSRC to change right
before the channels are bridged, which is what used to happen. It
seems that fixes were made to attempt limiting SSRC changes,
targeted mainly at sending DTMF. DTMF is not affecting the SSRC
with this change. There are two other control frames sent in
ast_channel_bridge that probably should also be changed to
AST_CONTROL_SRCCHANGE as well, but I'm going to leave this change
up to the discretion of resolving issue #17007. For reference -
old review implementing new control frame SRCCHANGE:
https://reviewboard.asterisk.org/r/540 (closes issue #17404)
Reported by: sdolloff Patches: bug17404.patch uploaded by jpeeler
(license 325) Tested by: sdolloff ........
2010-08-11 21:09 +0000 [r281763-281873] Leif Madsen <lmadsen@digium.com>
* configs/say.conf.sample, /: Merged revisions 281819 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r281819 | lmadsen | 2010-08-11 13:28:10 -0500 (Wed, 11
Aug 2010) | 6 lines Add Danish support to say.conf.sample (closes
issue #17836) Reported by: RoadKill Patches:
say.conf.sample.patch.dk uploaded by RoadKill (license 933)
........
* configs/say.conf.sample, /: Merged revisions 281762 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r281762 | lmadsen | 2010-08-11 12:51:40 -0500 (Wed, 11
Aug 2010) | 6 lines Allow say.conf to handle large numbers ending
with multiple zeros. (closes issue #17833) Reported by: RoadKill
Patches: say.conf.sample.patch.largenumbers uploaded by RoadKill
(license 933) ........
2010-08-11 15:17 +0000 [r281722] Tilghman Lesher <tlesher@digium.com>
* apps/app_readexten.c: Only set status TIMEOUT, if we have no
digits. (closes issue #15188) Reported by: jcovert Patches:
app_readexten.c.patch-1.6.2.8-rc1 uploaded by jcovert (license
551)
2010-08-10 18:04 +0000 [r281567-281574] Russell Bryant <russell@digium.com>
* main/sched.c: Don't move the time threshold for running scheduled
events on every iteration. Instead, only calculate the time
threshold each time ast_sched_runq() is called. (closes issue
#17742) Reported by: schmidts Patches: sched.c.patch uploaded by
schmidts (license 1077)
* apps/app_dial.c, /: Merged revisions 281566 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r281566 | russell | 2010-08-10 12:45:45 -0500 (Tue, 10 Aug 2010)
| 8 lines Reset visible indication after answer. (closes issue
#17641) Reported by: klaus3000 Patches:
ast1.6.2.9-app_dial-visible_indication.patch.txt uploaded by
klaus3000 (license 65) Tested by: schmidts ........
2010-08-09 20:46 +0000 [r281430] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: fixes SIP peers memory leak We zeroed out
the peer's addr before it was removed from the peers_by_ip
container. This made it impossible to be removed from the
container as the addr is the key used by the container to find
the peer. (closes issue #17774) Reported by: kkm Patches:
017774-sip-peer-leak-1.6.2.10.diff uploaded by kkm (license 888)
017774-sip-peer-leak-1.8.diff uploaded by kkm (license 888)
2010-08-09 20:07 +0000 [r281391] Jeff Peeler <jpeeler@digium.com>
* channels/chan_local.c, /: Merged revisions 281390 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r281390 | jpeeler | 2010-08-09 15:04:30 -0500 (Mon, 09
Aug 2010) | 13 lines Prevent loss of Caller ID information set on
local channel after masquerade. Caller ID set on the channel
before a masquerade occurs when using a local channel would cause
the information to be lost. The problem was that the information
was set on a channel destined to be hung up. The somewhat
confusing fix is to detect if any Caller ID has been set on the
channel and if so preswap the Caller ID data so that basically
the masquerade puts the data back. (closes issue #17138) Reported
by: kobaz Review: https://reviewboard.asterisk.org/r/847/
........
2010-08-05 13:11 +0000 [r281051] Russell Bryant <russell@digium.com>
* main/cdr.c: Cleanup default option value handling for cdr.conf
[general]. The default values would differ depending on whether
or not cdr.conf exists. That is no longer the case. Apply a
default value to the unanswered option. Define all default values
as named constants.
2010-08-05 07:40 +0000 [r280983] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/pbx.h, main/pbx.c, /: Merged revisions 280982
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r280982 | tilghman | 2010-08-05 02:28:33 -0500 (Thu, 05 Aug 2010)
| 8 lines Change context lock back to a mutex, because
functionality depends upon the lock being recursive. (closes
issue #17643) Reported by: zerohalo Patches:
20100726__issue17643.diff.txt uploaded by tilghman (license 14)
Tested by: zerohalo ........
2010-08-03 20:52 +0000 [r280671-280812] Tilghman Lesher <tlesher@digium.com>
* funcs/func_callerid.c, channels/chan_dahdi.c, /: Merged revisions
280811 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r280811 | tilghman | 2010-08-03 15:49:10 -0500 (Tue, 03 Aug 2010)
| 9 lines Prevent DAHDI channels from overriding the callerid,
once it's been set by the user. (closes issue #16661) Reported
by: jstapleton Patches: 20100414__issue16661.diff.txt uploaded by
tilghman (license 14) 20100415__issue16661__1.6.2.diff.txt
uploaded by tilghman (license 14) Tested by: jstapleton ........
* doc/asterisk.sgml, doc/asterisk.8, doc/Makefile (added): Document
-B and -W flags and regenerate manpage from sgml
* apps/app_voicemail.c: Allow the pipe, but also allow the comma
2010-08-02 21:14 +0000 [r280669] Jeff Peeler <jpeeler@digium.com>
* channels/chan_sip.c: Change SIP NOTIFY requests to expect a
response so authentication will work. This changes the request to
be sent with the transmit type XMIT_RELIABLE so that sip_ack
doesn't return false and cause the 401 to be ignored in cases
where authentication is required. (closes issue #14255) Reported
by: zktech
2010-07-29 21:07 +0000 [r280556] Tilghman Lesher <tlesher@digium.com>
* res/res_config_curl.c: Off-by-one error (closes issue #17590)
Reported by: atis Patches: 20100729__issue17590.diff.txt uploaded
by tilghman (license 14)
2010-07-29 20:42 +0000 [r280449-280551] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: fixes wrong SRV query for TLS connection
(closes issue #17612) Reported by: marcelloceschia Patches:
chan-sip_srvQuery.patch uploaded by marcelloceschia (license
1079) chan-sip_Trunk_srvQuery.patch uploaded by st (license 907)
chan-sip_asterisk18b1_srvQuery.patch uploaded by marcelloceschia
(license 1079) Tested by: marcelloceschia, st, pabelanger
* main/channel.c, /: Merged revisions 280448 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r280448 | dvossel | 2010-07-29 14:04:23 -0500 (Thu, 29 Jul 2010)
| 12 lines fixes issue with translator frame not getting freed A
translator frame even if it local storage so the translation path
can be freed. This issue prevented g729 licenses from being freed
up. (closes issue #17630) Reported by: manvirr Patches:
encoder_fix.diff uploaded by dvossel (license 671) Tested by:
manvirr, dvossel ........
2010-07-29 16:01 +0000 [r280345] Jean Galarneau <jgalarneau@digium.com>
* /, apps/app_meetme.c: Merged revisions 280341 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r280341 | jeang | 2010-07-29 10:52:31 -0500 (Thu, 29 Jul 2010) |
2 lines Fix a dsp structure leak occuring when a local channel is
put into a meetme conference, then masquaraded away. ABE-2422
........
2010-07-29 13:45 +0000 [r280306] Matthew Nicholson <mnicholson@digium.com>
* channels/chan_local.c: Implement support for
ast_channel_queryoption on local channels. Currently only
AST_OPTION_T38_STATE is supported. ABE-2229 Review:
https://reviewboard.asterisk.org/r/813/
2010-07-28 20:02 +0000 [r280231] Jason Parker <jparker@digium.com>
* sounds/Makefile: Work around some silly behavior on BSD. A
non-zero exit from a subshell should make the build fail. (closes
issue #17621)
2010-07-28 19:57 +0000 [r280229] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: Add missing enum value "unknown" to the
SS7 called_nai and calling_nai config options.
2010-07-28 19:54 +0000 [r280193-280227] Jason Parker <jparker@digium.com>
* build_tools/sha1sum-sh (added): Add sha1sum-sh in case there is
no util on the system.
* sounds/Makefile: Remove unnecessary subshells. Attempt to make
checksumming work. Also improves readability. (issue #17621)
Reported by: bjm Review: https://reviewboard.asterisk.org/r/808/
2010-07-28 16:51 +0000 [r280160] Sean Bright <sean@malleable.com>
* apps/app_queue.c: Plug a reference leak in app_queue when adding
members dynamically. (closes issue #17738) Reported by:
bobwienholt Patches: issue17738.patch uploaded by bobwienholt
(license 950) Tested by: bobwienholt, seanbright
2010-07-28 13:51 +0000 [r280089] Leif Madsen <lmadsen@digium.com>
* contrib/scripts/live_ast, /: Merged revisions 280088 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r280088 | lmadsen | 2010-07-28 08:50:38 -0500 (Wed, 28
Jul 2010) | 1 line Update help text to be less confusing.
........
2010-07-27 20:54 +0000 [r279946] David Vossel <dvossel@digium.com>
* main/audiohook.c, main/channel.c, /,
include/asterisk/audiohook.h: Merged revisions 279945 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r279945 | dvossel | 2010-07-27 15:33:40 -0500 (Tue, 27 Jul 2010)
| 19 lines remove empty audiohook write list on channel If a
channel has an audiohook write list created on it, that list
stays on the channel until the channel is destroyed. There is no
reason to keep that list on the channel if it becomes empty. If
it is empty that just means we are doing needless translating for
every ast_read and ast_write. This patch removes the audiohook
list from the channel once it is detected to be empty on either a
read or write. If a audiohook is added back to the channel after
this list is destroyed, the list just gets recreated as if it
never existed to begin with. (closes issue #17630) Reported by:
manvirr Review: https://reviewboard.asterisk.org/r/799/ ........
2010-07-27 17:54 +0000 [r279849-279883] Jason Parker <jparker@digium.com>
* makeopts.in, configure, configure.ac: Add SHA1SUM to configure,
since we require it for sounds/
* sounds/Makefile: Remove aptly-named EMPTY and BS vars, since they
aren't used anymore.
* sounds/Makefile: Simply sounds/Makefile some more.
2010-07-27 15:13 +0000 [r279784] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Fix bad behavior of dynamic_exclude_static
option in sip.conf. We were attempting to create a contactdeny
rule based on the peer's IP address before the peer's IP address
had been set. By moving the processing further down in the
function, we can ensure stuff works as we expect for it to.
(closes issue #17717) Reported by: mmichelson Patches:
17717.patch uploaded by mmichelson (license 60) Tested by:
DennisD
2010-07-26 22:59 +0000 [r279657] Jason Parker <jparker@digium.com>
* sounds/Makefile (added), sounds/Makefile.380 (removed),
configure, include/asterisk/autoconfig.h.in, sounds/Makefile.381
(removed), configure.ac: Really fix sounds Makefile (and make it
readableish). There was a rather large syntax error that should
have caused ALL versions of GNU make to fail. I don't know how it
worked.
2010-07-26 21:18 +0000 [r279609] Tilghman Lesher <tlesher@digium.com>
* configure, configure.ac: Dunno why this worked on my machine, but
it works better this way.
2010-07-26 20:25 +0000 [r279597] Gavin Henry <ghenry@suretecsystems.com>
* res/res_config_ldap.c: Apply all patches in:
https://issues.asterisk.org/view.php?id=13573 (closes issue
#13573) Reported by: navkumar Patches:
res_config_ldap-category.diff uploaded by navkumar (license 580)
res_config_ldap.patch uploaded by bencer (license 961)
res_config_ldap uploaded by bencer (license 961) Tested by:
suretec
2010-07-26 19:15 +0000 [r279561] Tilghman Lesher <tlesher@digium.com>
* sounds/Makefile (removed), configure, sounds/Makefile.380
(added), sounds/Makefile.381 (added), configure.ac: Use a special
Makefile for noobs who still have GNU Make 3.80. (Closes issue
#17716) Reported by: farisraouf
2010-07-26 15:41 +0000 [r279501] Sean Bright <sean@malleable.com>
* autoconf/ast_ext_lib.m4: Expand the correct value within
AST_OPTION_ONLY. (closes issue #17703) Reported by: stuarth
2010-07-24 23:58 +0000 [r279347] Bradley Latus <brad.latus@gmail.com>
* doc/asterisk.8: Minor update to man page
2010-07-23 22:11 +0000 [r279207] Richard Mudgett <rmudgett@digium.com>
* apps/app_queue.c, apps/app_dial.c, /: Merged revisions 279206 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r279206 | rmudgett | 2010-07-23 16:56:44 -0500 (Fri, 23 Jul 2010)
| 7 lines SIP promiscuous redirect could fail to dial the
redirect. The ast_channel was created with one variable to
ast_request() but the call to ast_call() that initiates the
outgoing call was using a different variable. The two variables
are not equivalent if the call_forward string included a channel
technology specifier. e.g., SIP/200 ........
2010-07-23 18:29 +0000 [r279112] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Backport sip_uri_params_cmp() fix from trunk
to 1.6.2.
2010-07-23 18:22 +0000 [r279072-279088] Russell Bryant <russell@digium.com>
* /: remove old properties
* /: Add branch-1.4-merged and branch-1.4-blocked properties to
1.6.2 branch.
2010-07-23 17:06 +0000 [r278983-278986] Tilghman Lesher <tlesher@digium.com>
* autoconf/ast_check_pwlib.m4, /, configure, configure.ac: Merged
revisions 278985 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r278985 | tilghman | 2010-07-23 12:05:16 -0500 (Fri, 23 Jul 2010)
| 12 lines Merged revisions 278984 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r278984 | tilghman | 2010-07-23 12:04:15 -0500 (Fri, 23 Jul 2010)
| 5 lines Establish a maximum version for openh323 (i.e. not
opal), because chan_h323 will fail to load, even if it links.
(issue #17679) Reported by: am ........ ................
* main/asterisk.c, /: Merged revisions 278982 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r278982 | tilghman | 2010-07-23 11:43:34 -0500 (Fri, 23 Jul 2010)
| 15 lines Merged revisions 278981 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r278981 | tilghman | 2010-07-23 11:42:25 -0500 (Fri, 23 Jul 2010)
| 8 lines Avoid race with consolethread on shutdown (on parallel
processors). (closes issue #17080) Reported by: sybasesql
Patches: 20100721__issue17080.diff.txt uploaded by tilghman
(license 14) Tested by: sybasesql ........ ................
2010-07-23 15:23 +0000 [r278934] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* channels/chan_dahdi.c: Two more typos to cancell.
2010-07-22 19:52 +0000 [r278709] Jeff Peeler <jpeeler@digium.com>
* main/xmldoc.c, /: Merged revisions 278708 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r278708 |
jpeeler | 2010-07-22 14:45:30 -0500 (Thu, 22 Jul 2010) | 16 lines
Add method for finding XML doc files for systems that don't
support GLOB_BRACE. In particular, Solaris and perhaps others do
not support the above mentioned GNU extension. In this case the
paths are simply expanded without the braces and the calls to
glob are made separately. Note: I could not explain memory
allocation failures that were being reported from within libxml
itself when making calls to glob without using GLOB_NOCHECK. This
is the only reason why that flag is being used. (closes issue
#15402) Reported by: snuffy Patches: bug_xmlpatt-v3.diff uploaded
by snuffy (license 35), modified by me ........
2010-07-22 19:32 +0000 [r278703] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: DNID does not get cleard on a new call
when using immediate=yes with ISDN signaling. When you are using
chan_dahdi ISDN signaling with immediate=yes and a call comes in
without a DNID then you get the DNID of a previous call.
Chan_dahdi does not touch the DNID field on a new call if it does
not have a DNID. Made always copy the DNID from the new call. The
patches backport the relevant changes from trunk -r210387.
(closes issue #17568) Reported by: wuwu Patches:
issue17568_v1.4.patch uploaded by rmudgett (license 664)
issue17568_v1.6.2.patch uploaded by rmudgett (license 664)
2010-08-10 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.11 Released.
2010-07-26 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.11-rc2 Released.
2010-07-26 Leif Madsen <lmadsen@digium.com>
* qwell, asterisk, branch-1.6.2, r279657 ***
Really fix sounds Makefile (and make it readableish).
There was a rather large syntax error that should have
caused ALL versions of GNU make to fail.
I don't know how it worked.
(Closes issue #17716)
2010-07-22 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.11-rc1 Released.
2010-07-22 15:00 +0000 [r278621] Mark Michelson <mmichelson@digium.com>
* main/channel.c, /: Merged revisions 278620 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r278620 | mmichelson | 2010-07-22 09:58:01 -0500 (Thu, 22 Jul
2010) | 19 lines Merged revisions 278618 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r278618 | mmichelson | 2010-07-22 09:55:04 -0500 (Thu, 22 Jul
2010) | 13 lines Allow PLC to function properly when channels use
SLIN for audio. If a channel involved in a bridge was using SLIN
audio, then translation paths were not guaranteed to be set up
properly since in all likelihood the number of translation steps
was only 1. This patch enforces the transcode_via_slin behavior
if transcode_via_slin or generic_plc is enabled and one of the
formats to make compatible is SLIN. AST-352 ........
................
2010-07-21 18:22 +0000 [r278524] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* channels/chan_dahdi.c: Fix invalid test for rxisoffhook in FXO
channels This fixes some cases of no outgoing calls on FXO before
an incoming call. Remove an unnecessary testing of an "off-hook"
bit from DAHDI for FXO (KS/GS) channels.In some cases the bit
would not be initialized properly before the first inbound call
and thus prevent an outgoing call. If those tests are actually
required by anybody, they should define DAHDI_CHECK_HOOKSTATE in
channels/sig_analog.c . (closes issue #14577) Reported by: jkroon
Patches: asterisk_chan_dahdi_hookstate_fix.diff uploaded by frawd
(license 610) Tested by: frawd Review:
https://reviewboard.asterisk.org/r/699/
2010-07-21 16:20 +0000 [r278479] Russell Bryant <russell@digium.com>
* /, res/res_timing_pthread.c: Merged revisions 278465 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r278465 | russell | 2010-07-21 11:15:00 -0500 (Wed, 21 Jul 2010)
| 41 lines Use poll() instead of select() in res_timing_pthread
to avoid stack corruption. This code did not properly check
FD_SETSIZE to ensure that it did not try to select() on fds that
were too large. Switching to poll() removes the limitation on the
maximum fd value. (closes issue #15915) Reported by: keiron
(closes issue #17187) Reported by: Eddie Edwards (closes issue
#16494) Reported by: Hubguru (closes issue #15731) Reported by:
flop (closes issue #12917) Reported by: falves11 (closes issue
#14920) Reported by: vrban (closes issue #17199) Reported by:
aleksey2000 (closes issue #15406) Reported by: kowalma (closes
issue #17438) Reported by: dcabot (closes issue #17325) Reported
by: glwgoes (closes issue #17118) Reported by: erikje possibly
other issues, too ... ........
2010-07-21 15:58 +0000 [r278025-278464] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_meetme.c: Merged revisions 278463 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r278463 |
tilghman | 2010-07-21 10:56:05 -0500 (Wed, 21 Jul 2010) | 11
lines Ensure realtime conferences are treated the same as static
conferences when trying to find an empty one. Also, parse the
useropts properly, when retrieving from realtime, and add them to
the existing flags. (closes issue #17502) Reported by: kenji
Patches: 20100720__issue17502.diff.txt uploaded by tilghman
(license 14) Tested by: kenji ........
* apps/app_voicemail.c, /: Merged revisions 278275 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r278275 | tilghman | 2010-07-20 17:40:19 -0500
(Tue, 20 Jul 2010) | 14 lines Merged revisions 278261 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r278261 | tilghman | 2010-07-20 17:23:13 -0500 (Tue, 20 Jul 2010)
| 7 lines Delete IMAP messages in reverse order, to ensure
reordering after each expunge does not cause deletion of the
wrong message. (closes issue #16350) Reported by: noahisaac
Patches: 20100623__issue16350.diff.txt uploaded by tilghman
(license 14) ........ ................
* main/autoservice.c, /, main/features.c,
include/asterisk/channel.h: Merged revisions 278272 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r278272 | tilghman | 2010-07-20 17:26:23 -0500
(Tue, 20 Jul 2010) | 11 lines Merged revisions 278167 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r278167 | tilghman | 2010-07-20 15:59:06 -0500 (Tue, 20 Jul 2010)
| 4 lines Do not queue up DTMF frames while a call is on hold.
(Fixes ABE-2110) ........ ................
* main/manager.c, /: Merged revisions 278024 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r278024 | tilghman | 2010-07-20 11:50:11 -0500 (Tue, 20 Jul 2010)
| 14 lines Merged revisions 278023 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r278023 | tilghman | 2010-07-20 11:37:18 -0500 (Tue, 20 Jul 2010)
| 7 lines Off-by-one error (closes issue #16506) Reported by:
nik600 Patches: 20100629__issue16506.diff.txt uploaded by
tilghman (license 14) ........ ................
2010-07-19 21:21 +0000 [r277966] Jean Galarneau <jgalarneau@digium.com>
* /, main/features.c: Merged revisions 277945 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r277945 | jeang | 2010-07-19 16:07:08 -0500 (Mon, 19 Jul 2010) |
15 lines Merged revisions 277906 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r277906 | jeang | 2010-07-19 15:16:36 -0500 (Mon, 19 Jul 2010) |
7 lines Avoid trying to pickup a parked extension before the park
operation is completed. A crash could occur if the extension is
picked up while the parking extension is being announced. Testing
pu->notquiteyet while searching for a parked extension resolves
this crash. (ABE-2418) ........ ................
2010-07-17 17:52 +0000 [r277774-277777] Tilghman Lesher <tlesher@digium.com>
* res/res_config_pgsql.c: Merge issues...
* /, autoconf/ast_func_fork.m4, configure,
include/asterisk/autoconfig.h.in: Merged revisions 277775 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r277775 | tilghman | 2010-07-17 12:42:32 -0500
(Sat, 17 Jul 2010) | 12 lines Merged revisions 277738 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r277738 | tilghman | 2010-07-17 11:59:11 -0500 (Sat, 17 Jul 2010)
| 5 lines Remove uclibc cross-compile triplet, as uclibc has a
working fork()... it's only uclinux that does not. (closes issue
#17616) Reported by: pprindeville ........ ................
* res/res_config_pgsql.c, res/res_config_odbc.c, /: Merged
revisions 277773 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r277773 | tilghman | 2010-07-17 12:39:28 -0500 (Sat, 17 Jul 2010)
| 15 lines Merged revisions 277568 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r277568 | tilghman | 2010-07-16 16:54:29 -0500 (Fri, 16 Jul 2010)
| 8 lines Since we split values at the semicolon, we should store
values with a semicolon as an encoded value. (closes issue
#17369) Reported by: gkservice Patches:
20100625__issue17369.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman ........ ................
2010-07-16 23:37 +0000 [r277666] Tim Ringenbach <tim.ringenbach@gmail.com>
* /, main/features.c: Merged revisions 277657 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r277657 | tringenbach | 2010-07-16 18:23:15 -0500 (Fri, 16 Jul
2010) | 16 lines Merged revisions 277625 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r277625 | tringenbach | 2010-07-16 17:43:39 -0500 (Fri, 16 Jul
2010) | 9 lines Save and restore AST_FLAG_BRIDGE_HANGUP_DONT on
attended transfer. ast_bridge_call() clears
AST_FLAG_BRIDGE_HANGUP_DONT. But during an attended transfer,
ast_bridge_call() is called for a second bridge on the same
channel, and it clears that flag, which still needs to get set
for when the original ast_bridge_call() gets control back and
checks it. Review: https://reviewboard.asterisk.org/r/741
........ ................
2010-07-16 21:31 +0000 [r277563] Matthew Nicholson <mnicholson@digium.com>
* /, channels/chan_sip.c: Merged revisions 277530 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r277530 | mnicholson | 2010-07-16 16:24:45 -0500 (Fri, 16 Jul
2010) | 11 lines Merged revisions 277497 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r277497 | mnicholson | 2010-07-16 16:18:38 -0500 (Fri, 16 Jul
2010) | 4 lines Default to no udptl error correction so that
error correction will be disabled in the event that the remote
end indicates that they do not support the error correction mode
we requested. FAX-128 ........ ................
2010-07-16 21:16 +0000 [r277489] Jeff Peeler <jpeeler@digium.com>
* apps/app_queue.c, /: Merged revisions 277488 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r277488 |
jpeeler | 2010-07-16 16:16:08 -0500 (Fri, 16 Jul 2010) | 10 lines
Fix reporting estimated queue hold time. Just say the number of
seconds (after minutes) rather than doing some incorrect
calculation with respect to minutes. (closes issue #17498)
Reported by: corruptor Patches: holdesecs_bug.diff uploaded by
corruptor (license 253) ........
2010-07-16 20:35 +0000 [r277485] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 277467 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r277467 | rmudgett | 2010-07-16 15:27:51 -0500
(Fri, 16 Jul 2010) | 22 lines Merged revisions 277419 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r277419 | rmudgett | 2010-07-16 15:18:54 -0500 (Fri, 16 Jul 2010)
| 15 lines priexclusive in chan_dahdi.conf ignored when reloading
dahdi module During a reload, the priexclusive and outsignalling
parameters are not read in from the config file as intended.
Unfortunately, they get set to defaults as a result. This patch
makes sure that they do not get set to defaults during a reload.
(closes issue #17441) Reported by: mtryfoss Patches:
issue17441_v1.4.patch uploaded by rmudgett (license 664)
issue17441_v1.6.2.patch uploaded by rmudgett (license 664)
issue17441_trunk.patch uploaded by rmudgett (license 664) Tested
by: rmudgett ........ ................
2010-07-16 20:30 +0000 [r277478] Tilghman Lesher <tlesher@digium.com>
* res/res_musiconhold.c, contrib/realtime/mysql/musiconhold.sql
(added), /: Merged revisions 277452 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r277452 |
tilghman | 2010-07-16 15:25:11 -0500 (Fri, 16 Jul 2010) | 2 lines
Add documentation for MOH realtime fields ........
2010-07-16 19:24 +0000 [r277377] Jeff Peeler <jpeeler@digium.com>
* apps/app_queue.c, /: Merged revisions 277366 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r277366 |
jpeeler | 2010-07-16 14:22:49 -0500 (Fri, 16 Jul 2010) | 7 lines
Add missing handling for ringing state for use with queue empty
options. (closes issue #17471) Reported by: jazzy Patches:
app_queue.c.diff uploaded by jazzy (license 1056) ........
2010-07-16 18:33 +0000 [r277338] Matthew Nicholson <mnicholson@digium.com>
* main/pbx.c, /: Merged revisions 277331 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r277331 | mnicholson | 2010-07-16 13:31:08 -0500 (Fri, 16 Jul
2010) | 15 lines Merged revisions 277327 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r277327 | mnicholson | 2010-07-16 13:30:22 -0500 (Fri, 16 Jul
2010) | 8 lines Interpret device state AST_DEVICE_UNKNOWN as
extension state AST_EXTENSION_NOT_INUSE. (closes issue #16035)
Reported by: francesco_r Patches: pbx.c.patch uploaded by
viniciusfontes (license 978) Tested by: francesco_r, agx, lawbar
........ ................
2010-07-16 18:14 +0000 [r277264] Tilghman Lesher <tlesher@digium.com>
* main/manager.c, /: Merged revisions 277263 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r277263 | tilghman | 2010-07-16 13:14:05 -0500 (Fri, 16 Jul 2010)
| 12 lines Merged revisions 277261 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r277261 | tilghman | 2010-07-16 13:04:11 -0500 (Fri, 16 Jul 2010)
| 5 lines If variable gotten is not set, will segfault on
Solaris. (closes issue #17636) Reported by: bklang ........
................
2010-07-16 17:31 +0000 [r277256] Matthew Nicholson <mnicholson@digium.com>
* main/channel.c, /: Merged revisions 277250 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r277250 | mnicholson | 2010-07-16 12:30:39 -0500 (Fri, 16 Jul
2010) | 11 lines Merged revisions 277247 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r277247 | mnicholson | 2010-07-16 12:29:57 -0500 (Fri, 16 Jul
2010) | 4 lines For pass through DTMF tones, measure the actual
duration between the begin and end packets on the wire. If it is
detected to be less than AST_MIN_DTMF_DURATION, trigger dtmf
emulation. AST-362 ........ ................
2010-07-16 17:18 +0000 [r277188] Paul Belanger <paul.belanger@polybeacon.com>
* /, apps/app_amd.c: Merged revisions 277183 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r277183 | pabelanger | 2010-07-16 13:13:46 -0400 (Fri, 16 Jul
2010) | 15 lines Merged revisions 277182 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r277182 | pabelanger | 2010-07-16 13:10:36 -0400 (Fri, 16 Jul
2010) | 8 lines Total analysis time error with SIP and silence
suppression When using app_amd with SIP providers that have
silence suppression on, the iTotalTime count increases
exponentially. (closes issue #17656) Reported by: juls ........
................
2010-07-16 15:21 +0000 [r277144] Sean Bright <sean@malleable.com>
* /, main/translate.c: Merged revisions 277143 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r277143 |
seanbright | 2010-07-16 11:20:40 -0400 (Fri, 16 Jul 2010) | 8
lines Avoid crashing when installing a duplicate translation path
with a lower cost. (closes issue #17092) Reported by: moy
Patches: translate.rev254273.patch uploaded by moy (license 222)
Tested by: moy ........
2010-07-15 20:42 +0000 [r276572-276809] Jeff Peeler <jpeeler@digium.com>
* /, channels/chan_sip.c: Merged revisions 276788 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r276788 |
jpeeler | 2010-07-15 15:21:03 -0500 (Thu, 15 Jul 2010) | 6 lines
Correct not setting the bindport before attempting to open the
socket. Related to changes from 276571, I was accidentally
testing with a port set in my configuration causing me to miss
this. Also moved the TCP handling as well to occur before
build_peer is called. ........
* main/channel.c, /: Merged revisions 276653 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r276653 | jpeeler | 2010-07-15 08:51:11 -0500 (Thu, 15 Jul 2010)
| 9 lines Merged revisions 276652 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r276652 | jpeeler | 2010-07-15 08:48:58 -0500 (Thu, 15 Jul 2010)
| 2 lines In a perfect world, the frame source would never be
NULL. In the meantime, don't crash when it is. ........
................
* /, channels/chan_sip.c: Merged revisions 276571 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r276571 |
jpeeler | 2010-07-14 17:58:24 -0500 (Wed, 14 Jul 2010) | 21 lines
Fix MWI notification transmission problems over SIP. MWI updates
were not being sent if no messages were found in the event cache.
This was corrected since a phone may need to clear its MWI status
configured previously from another mailbox. Upon module or sip
reload, MWI updates could not be sent due to the sipsock socket
not being set early enough in reload_config. The code handling
the descriptor assignment and such has simply been moved before
the call to build_peer. Issuing a sip reload cleared the IP
address of the peer, but skipped checking the database for
registration information. The database is now checked both for
sip reload and actually reloading the module. If a transmission
occurs before the do_monitor thread has started, do not attempt
to send a signal to it. (closes issue #17398) Reported by: ip-rob
........
2010-07-14 20:16 +0000 [r276442] Kevin P. Fleming <kpfleming@digium.com>
* main/loader.c, /: Merged revisions 276441 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r276441 |
kpfleming | 2010-07-14 15:15:48 -0500 (Wed, 14 Jul 2010) | 4
lines Don't try to call an embedded module's backup_globals()
function until after confirming it exists. ........
2010-07-14 11:52 +0000 [r276269] Leif Madsen <lmadsen@digium.com>
* /, configs/voicemail.conf.sample: Merged revisions 276268 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r276268 | lmadsen | 2010-07-14 06:51:48 -0500
(Wed, 14 Jul 2010) | 9 lines Merged revisions 276267 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r276267 | lmadsen | 2010-07-14 06:49:01 -0500 (Wed, 14
Jul 2010) | 1 line Update documentation for voicemail.conf
externpass option. ........ ................
2010-07-13 19:11 +0000 [r276125] Russell Bryant <russell@digium.com>
* /, main/features.c: Merged revisions 276124 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r276124 | russell | 2010-07-13 14:09:42 -0500 (Tue, 13 Jul 2010)
| 9 lines Merged revisions 276123 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r276123 | russell | 2010-07-13 14:06:53 -0500 (Tue, 13 Jul 2010)
| 2 lines Use chan->cdr instead of chan_cdr (just like peer->cdr
instead of peer_cdr in the last commit). ........
................
2010-07-13 19:01 +0000 [r276121] Jeff Peeler <jpeeler@digium.com>
* /, apps/app_meetme.c: Merged revisions 276074 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r276074 | jpeeler | 2010-07-13 12:37:40 -0500 (Tue, 13 Jul 2010)
| 19 lines Merged revisions 275773 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r275773 | jpeeler | 2010-07-12 15:34:51 -0500 (Mon, 12 Jul 2010)
| 12 lines Make user removals and traversals thread safe in
meetme. Race conditions present in meetme involving the user list
where a lack of locking has the potential for a user to be
removed during a traversal or as in the case of the reporter
after checking if the list is empty could cause a crash. Fixing
this was done by convering the userlist to an ao2 container.
(closes issue #17390) Reported by: Vince Review:
https://reviewboard.asterisk.org/r/746/ ........ ................
2010-07-13 16:55 +0000 [r275996] Russell Bryant <russell@digium.com>
* /, main/features.c: Merged revisions 275995 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r275995 | russell | 2010-07-13 11:53:44 -0500 (Tue, 13 Jul 2010)
| 21 lines Merged revisions 275994 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r275994 | russell | 2010-07-13 11:51:18 -0500 (Tue, 13 Jul 2010)
| 14 lines Access peer->cdr directly instead of through a saved
off reference. At this point in the code, it is possible that
peer_cdr may be invalid. Specifically, in the blind transfer
code, CDRs are swapped between channels. So, peer_cdr is no
longer == peer->cdr. The scenario that exposed a crash in this
code was a blind transfer that hit the system call limit, causing
the transferee channel to get destroyed after the transfer
attempt failed. Even if it succeeds and this code doesn't crash,
this code was still trying to reset a CDR on a channel that was
now owned by a different thread, which is a BadThing(tm).
(ABE-2417) ........ ................
2010-07-13 14:49 +0000 [r275911] Tilghman Lesher <tlesher@digium.com>
* contrib/realtime/mysql, contrib/realtime/oracle,
contrib/scripts/sip-friends.sql (removed),
contrib/realtime/mysql/sipfriends.sql,
contrib/realtime/mysql/voicemail.sql, contrib/scripts/vmdb.sql
(removed), contrib/realtime/mysql/meetme.sql,
contrib/realtime/sqlserver, contrib/scripts/realtime_pgsql.sql
(removed), contrib/scripts/iax-friends.sql (removed), /,
contrib/realtime/mysql/iaxfriends.sql, contrib/scripts/meetme.sql
(removed), contrib/realtime (added), contrib/realtime/postgresql,
contrib/realtime/postgresql/realtime.sql: Merged revisions 275910
via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r275910 | tilghman | 2010-07-13 09:48:40 -0500
(Tue, 13 Jul 2010) | 9 lines Merged revisions 275909 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r275909 | tilghman | 2010-07-13 09:47:30 -0500 (Tue, 13
Jul 2010) | 2 lines Move SQL scripts into their own
database-specific directories. ........ ................
2010-07-12 17:26 +0000 [r275706] Jeff Peeler <jpeeler@digium.com>
* main/channel.c, /: Merged revisions 275682 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r275682 | jpeeler | 2010-07-12 12:21:01 -0500 (Mon, 12 Jul 2010)
| 18 lines Merged revisions 275665 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r275665 | jpeeler | 2010-07-12 11:58:39 -0500 (Mon, 12 Jul 2010)
| 11 lines Change ast_write to not stop generator when called
from ast_prod. For SIP channels configured with the
progressinband option on, the ringback was being immediately
stopped. This problem was due to ast_prod being moved for a
deadlock fix in 259858. Prodding the channel after setting up the
generator triggered the check in ast_write to stop the generator.
The fix here should write the frame the same as was done before
the call to ast_prod was moved. (closes issue #17372) Reported
by: tech_admin ........ ................
2010-07-12 15:38 +0000 [r275627] Leif Madsen <lmadsen@digium.com>
* cdr/cdr_pgsql.c, /: Merged revisions 275626 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r275626 |
lmadsen | 2010-07-12 10:37:01 -0500 (Mon, 12 Jul 2010) | 11 lines
cdr_pgsql does not detect when a table is found. This change adds
an ERROR message to let you know when a failure exists to get the
columns from the pgsql database, which typically means that the
table does not exist. (closes issue #17478) Reported by: kobaz
Patches: cdr_pgsql.patch uploaded by kobaz (license 834) Tested
by: kobaz, russell, lmadsen ........
2010-07-10 15:11 +0000 [r275311-275469] Russell Bryant <russell@digium.com>
* configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions
245192 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r245192 |
mmichelson | 2010-02-06 08:43:03 -0600 (Sat, 06 Feb 2010) | 21
lines Remove useless sip options related to hash table size.
First off, these options weren't actually doing anything. By the
time the options were parsed, the peer and dialog containers had
already been allocated with their default values. Second, hash
table size is something that doesn't really make sense to change
in a config file. If a user is that interested in changing the
hashtable size, he can modify the source itself. I have removed
the parsing of the hash_peer, hash_user, and hash_dialog options.
I have removed the hash_user_size variable altogether since it is
not used at all. I also changed hash_peer_size and
hash_dialog_size to be constant, and have changed the symbols to
be in all caps as constants typically are. I have also removed
the entire section in sip.conf.sample regarding configurable
hashtable sizes. ........ (merge to 1.6.2 inspired by issue
#17553)
* /: unblock a rev
* configs/features.conf.sample, /, main/features.c: Merged
revisions 275424 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r275424 |
russell | 2010-07-09 16:57:21 -0500 (Fri, 09 Jul 2010) | 27 lines
Fix some issues related to dynamic feature groups in
features.conf. The bridge handling code did not properly consider
feature groups when setting parameters that would affect whether
or not a native bridge would be attempted. If DYNAMIC_FEATURES
only include a feature group, a native bridge would occur that
may prevent features from working. Fix a bug in verbose output
that would show the key mapping as empty if it was using the
default mapping and not a custom mapping in the feature group.
Add feature groups to the output of "features show". Adjust the
feature execution logic to match that of the logic when executing
a feature that was not configured through a feature group. Update
features.conf.sample to show that an '=' is still required if
using the default key mapping from [applicationmap]. Finally,
clean up a little bit of formatting to better coform to coding
guidelines while in the area. (closes issue #17589) Reported by:
lmadsen Patches: issue_17589.rev4.txt uploaded by russell
(license 2) Tested by: russell, lmadsen ........
* /, main/features.c: Merged revisions 275310 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r275310 |
russell | 2010-07-09 14:58:06 -0500 (Fri, 09 Jul 2010) | 2 lines
Add missing ao2_iterator_destroy(). ........
2010-07-09 19:23 +0000 [r275260] Paul Belanger <paul.belanger@polybeacon.com>
* /, channels/chan_sip.c: Merged revisions 275249 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r275249 | pabelanger | 2010-07-09 15:21:27 -0400 (Fri, 09 Jul
2010) | 15 lines Merged revisions 275241 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r275241 | pabelanger | 2010-07-09 15:20:00 -0400 (Fri, 09 Jul
2010) | 8 lines Fix logging message for stale nonce. (closes
issue #17582) Reported by: kenner Patches: chan_sip.c.diff
uploaded by kenner (license 1040) Tested by: lmadsen ........
................
2010-07-09 18:24 +0000 [r275191] Matthew Nicholson <mnicholson@digium.com>
* main/loader.c, /: Merged revisions 275186 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r275186 | mnicholson | 2010-07-09 13:24:03 -0500 (Fri, 09 Jul
2010) | 9 lines Merged revisions 275182 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r275182 | mnicholson | 2010-07-09 13:23:23 -0500 (Fri, 09 Jul
2010) | 2 lines give a better error message when attempting to
unload a module that is not loaded ........ ................
2010-07-09 18:11 +0000 [r275148] Russell Bryant <russell@digium.com>
* configs/features.conf.sample, /: Merged revisions 275147 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r275147 | russell | 2010-07-09 13:11:13 -0500 (Fri, 09
Jul 2010) | 2 lines Move parking lot sample config out from the
middle of dynamic features sample config. ........
2010-07-09 17:51 +0000 [r275029-275145] Matthew Nicholson <mnicholson@digium.com>
* main/loader.c, /: Merged revisions 275144 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r275144 | mnicholson | 2010-07-09 12:50:45 -0500 (Fri, 09 Jul
2010) | 9 lines Merged revisions 275143 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r275143 | mnicholson | 2010-07-09 12:50:05 -0500 (Fri, 09 Jul
2010) | 2 lines don't unload modules that returned
AST_MODULE_LOAD_DECLINE when they were loaded ........
................
* apps/app_dial.c, /: Merged revisions 275028 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r275028 | mnicholson | 2010-07-09 11:05:58 -0500 (Fri, 09 Jul
2010) | 15 lines Merged revisions 275027 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r275027 | mnicholson | 2010-07-09 11:04:21 -0500 (Fri, 09 Jul
2010) | 8 lines Clear the AST_CDR_FLAG_DIALED flag for channels
going into the pbx via the G option in app_dial (closes issue
#17592) Reported by: jamicque Patches: G-flag-cdr-fix1.diff
uploaded by mnicholson (license 96) Tested by: jamicque,
mnicholson ........ ................
2010-07-09 15:39 +0000 [r275023] Russell Bryant <russell@digium.com>
* include/asterisk/test.h, /, main/test.c: Merged revisions 275022
via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r275022 | russell | 2010-07-09 10:35:53 -0500
(Fri, 09 Jul 2010) | 11 lines Merged revisions 275021 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r275021 | russell | 2010-07-09 10:33:08 -0500 (Fri, 09 Jul 2010)
| 4 lines Document that a leading and trailing slash is expected
for test categories. Also, emit a warning if a test is registered
without one of these. ........ ................
2010-07-07 18:34 +0000 [r274627-274640] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 274639 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r274639 | rmudgett | 2010-07-07 13:32:35 -0500 (Wed, 07 Jul 2010)
| 1 line Add missing conditional around chan_dahdi
mfcr2_skip_category config parameter. ........
* channels/chan_dahdi.c, /: Merged revisions 274595 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r274595 | rmudgett | 2010-07-07 13:20:00 -0500
(Wed, 07 Jul 2010) | 9 lines Merged revisions 274579 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r274579 | rmudgett | 2010-07-07 13:12:41 -0500 (Wed, 07
Jul 2010) | 1 line Close the DAHDI FD on error when processing
chan_dahdi toneduration config parameter. ........
................
2010-07-07 06:16 +0000 [r274419] Tilghman Lesher <tlesher@digium.com>
* configs/say.conf.sample, /: Merged revisions 274418 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r274418 | tilghman | 2010-07-07 01:15:43 -0500
(Wed, 07 Jul 2010) | 15 lines Merged revisions 274417 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r274417 | tilghman | 2010-07-07 01:13:54 -0500 (Wed, 07 Jul 2010)
| 8 lines Correct how 100, 200, 300, etc. is said. Also add the
crazy British numbers. (closes issue #16102) Reported by: Delvar
Patches: say.conf.fix.patch uploaded by Delvar (license 908)
(plus a few additional fixes and simplifications by me) ........
................
2010-07-06 23:06 +0000 [r274360] Terry Wilson <twilson@digium.com>
* configs/sip.conf.sample, channels/chan_sip.c: Merged revisions
274284 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r274284 | twilson | 2010-07-06 17:15:27 -0500 (Tue, 06 Jul 2010)
| 18 lines Merged revisions 274280 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r274280 | twilson | 2010-07-06 17:08:20 -0500 (Tue, 06 Jul 2010)
| 9 lines Add option to not do a call forward on 482 Loop
Detected Asterisk has always set up a forwarded call when
receiving a 482 Loop Detected. This prevents handling the call
failure by just continuing on in the dialplan. Since this would
be a change in behavior, the new option to disable this behavior
is forwardloopdetected which defaults to 'yes'. Review:
https://reviewboard.asterisk.org/r/764/ ........ ................
2010-07-06 22:30 +0000 [r274347] Jeff Peeler <jpeeler@digium.com>
* configs/sip.conf.sample, /: Merged revisions 274316 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r274316 | jpeeler | 2010-07-06 17:23:35 -0500
(Tue, 06 Jul 2010) | 14 lines Merged revisions 274283 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r274283 | jpeeler | 2010-07-06 17:15:21 -0500 (Tue, 06 Jul 2010)
| 7 lines Correct sip.conf.sample comments for prematuremedia
option. (closes issue #17513) Reported by: festr Patches: patch
uploaded by festr (license 443) ........ ................
2010-07-06 22:10 +0000 [r274282] Tilghman Lesher <tlesher@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 274281 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r274281 | tilghman | 2010-07-06 17:09:23 -0500 (Tue, 06 Jul 2010)
| 2 lines Status shows all non-CRC4 lines as "yellow", even if
"yellow" was not in the bitfield. ........
2010-07-06 14:33 +0000 [r274168] Mark Michelson <mmichelson@digium.com>
* main/rtp.c, /: Merged revisions 274164 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r274164 | mmichelson | 2010-07-06 09:31:13 -0500 (Tue, 06 Jul
2010) | 22 lines Merged revisions 274157 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r274157 | mmichelson | 2010-07-06 09:29:23 -0500 (Tue, 06 Jul
2010) | 16 lines Fix problem with RFC 2833 DTMF not being
accepted. A recent check was added to ensure that we did not
erroneously detect duplicate DTMF when we received packets out of
order. The problem was that the check did not account for the
fact that the seqno of an RTP stream will roll over back to 0
after hitting 65535. Now, we have a secondary check that will
ensure that the seqno rolling over will not cause us to stop
accepting DTMF. (closes issue #17571) Reported by: mdeneen
Patches: rtp_seqno_rollover.patch uploaded by mmichelson (license
60) Tested by: richardf, maxochoa, JJCinAZ ........
................
2010-07-05 13:55 +0000 [r273888] Paul Belanger <paul.belanger@polybeacon.com>
* main/config.c, /: Merged revisions 273886 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r273886 | pabelanger | 2010-07-05 09:53:44 -0400 (Mon, 05 Jul
2010) | 15 lines Merged revisions 273884 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r273884 | pabelanger | 2010-07-05 09:51:29 -0400 (Mon, 05 Jul
2010) | 8 lines Remove extra line breaks from 'core show config
mappings' (closes issue #17583) Reported by: pabelanger Patches:
issue17583.patch uploaded by pabelanger (license 224) Tested by:
lmadsen ........ ................
2010-07-03 02:43 +0000 [r273716-273831] Tilghman Lesher <tlesher@digium.com>
* channels/chan_local.c, /, channels/chan_agent.c,
channels/chan_h323.c, include/asterisk/lock.h: Merged revisions
273830 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r273830 | tilghman | 2010-07-02 21:36:31 -0500 (Fri, 02 Jul 2010)
| 16 lines Merged revisions 273793 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r273793 | tilghman | 2010-07-02 16:36:39 -0500 (Fri, 02 Jul 2010)
| 9 lines Have the DEADLOCK_AVOIDANCE macro warn when an unlock
fails, to help catch potentially large software bugs. (closes
issue #17407) Reported by: pdf Patches:
20100527__issue17407.diff.txt uploaded by tilghman (license 14)
Review: https://reviewboard.asterisk.org/r/751/ ........
................
* main/autoservice.c, /: Merged revisions 273718 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r273718 | tilghman | 2010-07-02 12:10:59 -0500 (Fri, 02 Jul 2010)
| 15 lines Merged revisions 273717 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r273717 | tilghman | 2010-07-02 12:09:47 -0500 (Fri, 02 Jul 2010)
| 8 lines Autoservice loop optimization causes a busy loop, when
channels are serviced while in hangup. (closes issue #17564)
Reported by: ramonpeek Patches: 20100630__issue17564.diff.txt
uploaded by tilghman (license 14) Tested by: ramonpeek ........
................
* apps/app_queue.c, /: Merged revisions 273714 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r273714 |
tilghman | 2010-07-02 11:57:28 -0500 (Fri, 02 Jul 2010) | 2 lines
The switch fallthrough could create some errorneous situations,
so best to force directly to the default case. ........
2010-07-02 15:59 +0000 [r273642] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* channels/chan_iax2.c, apps/app_voicemail.c,
channels/chan_dahdi.c, channels/chan_sip.c, res/res_agi.c: Fix
typos reported by Lintian
2010-07-01 22:17 +0000 [r273571] Russell Bryant <russell@digium.com>
* main/datastore.c, /: Merged revisions 273566 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r273566 | russell | 2010-07-01 17:16:23 -0500 (Thu, 01 Jul 2010)
| 14 lines Merged revisions 273565 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r273565 | russell | 2010-07-01 17:09:19 -0500 (Thu, 01 Jul 2010)
| 7 lines Don't return a partially initialized datastore. If
memory allocation fails in ast_strdup(), don't return a partially
initialized datastore. Bad things may happen. (related to
ABE-2415) ........ ................
2010-07-01 20:29 +0000 [r273356-273529] Jeff Peeler <jpeeler@digium.com>
* /, apps/app_meetme.c: Merged revisions 273522 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r273522 | jpeeler | 2010-07-01 15:28:15 -0500 (Thu, 01 Jul 2010)
| 21 lines Merged revisions 273474 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r273474 | jpeeler | 2010-07-01 15:19:16 -0500 (Thu, 01 Jul 2010)
| 14 lines Allow admin user to join conference without using
admin mode and no user pin. Configuring the conference in
meetme.conf like the following: conf => 2345,,6666 did not prompt
for pin when used without admin mode. This meant that the
conference could not be joined as an admin even if the user knew
the correct pin. The original bug report was submitted claiming
that the blank user pin should deny entry into the conference. I
think a better way to handle this would be with a feature
enhancement that used the following syntax: conf => 2345,X,6666 -
where X denotes no acceptable pin allowed (closes issue #15704)
Reported by: modelnine ........ ................
* /, apps/app_meetme.c: Merged revisions 273355 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r273355 | jpeeler | 2010-07-01 10:12:31 -0500 (Thu, 01 Jul 2010)
| 19 lines Merged revisions 273354 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r273354 | jpeeler | 2010-07-01 10:05:43 -0500 (Thu, 01 Jul 2010)
| 12 lines Ensure channel placed in meetme in ringing state is
properly hung up. An outgoing channel placed in meetme while
still ringing which was then hung up would not exit meetme and
the channel was not properly destroyed. Specifically checking for
this scenario by looking at the appropriate control frames
resolves the issue. (closes issue #15871) Reported by: Ivan
Patches: meetme_congestion_trunk_v2.patch uploaded by Ivan
(license 229) ........ ................
2010-07-01 14:39 +0000 [r273271-273353] Matthew Nicholson <mnicholson@digium.com>
* main/manager.c, /: Merged revisions 273352 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r273352 |
mnicholson | 2010-07-01 09:37:37 -0500 (Thu, 01 Jul 2010) | 2
lines Fixed whitespace problems ........
* main/manager.c, /: Merged revisions 273350 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r273350 |
mnicholson | 2010-07-01 09:34:31 -0500 (Thu, 01 Jul 2010) | 2
lines Altered my comment about TCP_NODELAY ........
* main/manager.c, /: Merged revisions 273270 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r273270 |
mnicholson | 2010-06-30 13:48:21 -0500 (Wed, 30 Jun 2010) | 2
lines Set TCP_NODELAY on manager TCP sockets to prevent delays on
outgoing packets. This regression was introduced in r48338.
AST-359 ........
2010-06-30 17:32 +0000 [r273193-273234] Paul Belanger <paul.belanger@polybeacon.com>
* main/rtp.c, /: Merged revisions 273233 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r273233 |
pabelanger | 2010-06-30 13:28:04 -0400 (Wed, 30 Jun 2010) | 11
lines Fix rt(c)p set debug ip taking wrong argument Also clean up
some coding errors. (closes issue #17469) Reported by: wdoekes
Patches: astsvn-rtp-set-debug-ip.patch uploaded by wdoekes
(license 717) Tested by: wdoekes, pabelanger ........
* /: Revert previous commit; res_rtp_asterisk.c does not exist.
* /: Unblock revisions 218107 ........ r218107 | mvanbaak |
2009-09-12 15:08:16 +0200 (Sat, 12 Sep 2009) | 8 lines use the
actual given ip address for 'rtp set debug ip <foo>' instead of
the word 'ip' (closes issue 0015711) Reported by: davidw Patches:
2009082800-rtpdebug.diff.txt uploaded by mvanbaak (license 7)
Tested by: davidw ........
2010-06-30 01:07 +0000 [r273056-273145] Tilghman Lesher <tlesher@digium.com>
* main/manager.c, /: Merged revisions 273144 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r273144 |
tilghman | 2010-06-29 20:07:02 -0500 (Tue, 29 Jun 2010) | 8 lines
Permission checking for the system application is backwards.
(closes issue #17550) Reported by: kenner Patches: manager.c.diff
uploaded by kenner (license 1040) Tested by: kenner ........
* main/config.c, /: Merged revisions 273142 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r273142 |
tilghman | 2010-06-29 20:01:14 -0500 (Tue, 29 Jun 2010) | 5 lines
Don't attempt to proceed if our internal parser indicates an
invalid file. (closes issue #17560) Reported by: Nick_Lewis
........
* /, channels/chan_sip.c: Merged revisions 273078 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r273078 | tilghman | 2010-06-29 18:20:40 -0500 (Tue, 29 Jun 2010)
| 17 lines Merged revisions 273060 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r273060 | tilghman | 2010-06-29 18:15:28 -0500 (Tue, 29 Jun 2010)
| 10 lines Allow the "useragent" value to be restored into memory
from the realtime backend. This value is purely informational. It
does not alter configuration at all. (closes issue #16029)
Reported by: Guggemand Patches: realtime-useragent.patch uploaded
by Guggemand (license 897) Tested by: Guggemand ........
................
* main/channel.c, /: Merged revisions 273058 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r273058 | tilghman | 2010-06-29 17:59:51 -0500 (Tue, 29 Jun 2010)
| 11 lines Recorded merge of revisions 273057 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r273057 | tilghman | 2010-06-29 17:58:58 -0500 (Tue, 29 Jun 2010)
| 4 lines _Really_ skip the channel... don't just retry for
another 200 cycles. (Closes issue SWP-1652, ABE-2240) ........
................
* main/pbx.c, /: Merged revisions 273054 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r273054 |
tilghman | 2010-06-29 17:39:22 -0500 (Tue, 29 Jun 2010) | 11
lines Send DialPlanComplete as a response, not as a separate
event. Otherwise, it goes to all manager sessions and may exclude
the current session, if the Events mask excludes it. (closes
issue #17504) Reported by: rrb3942 Patches:
showdialplan_patch.diff uploaded by rrb3942 (license 1003) Tested
by: rrb3942 ........
2010-06-29 16:43 +0000 [r272972] Russell Bryant <russell@digium.com>
* main/asterisk.c, /: Merged revisions 253357 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r253357 |
russell | 2010-03-18 13:18:43 -0500 (Thu, 18 Mar 2010) | 8 lines
Increase CLI command output timeout for asterisk -rx to 60
seconds. (closes issue #17049) Reported by: russell Tested by:
russell Review: https://reviewboard.asterisk.org/r/573/ ........
2010-07-22 Leif Madsen <lmadsen@digium.com>
* Release Asterisk 1.6.2.10
* Included a fix for res_timing_pthread per the description below:
r278465 | russell | 2010-07-21 11:15:00 -0500 (Wed, 21 Jul 2010) | 41 lines
Use poll() instead of select() in res_timing_pthread to avoid stack corruption.
This code did not properly check FD_SETSIZE to ensure that it did not try to
select() on fds that were too large. Switching to poll() removes the limitation
on the maximum fd value.
2010-07-07 Leif Madsen <lmadsen@digium.com>
* Release Asterisk 1.6.2.10-rc2
* Fix problem with RFC 2833 DTMF not being accepted.
A recent check was added to ensure that we did not erroneously
detect duplicate DTMF when we received packets out of order.
The problem was that the check did not account for the fact that
the seqno of an RTP stream will roll over back to 0 after hitting
65535. Now, we have a secondary check that will ensure that the
seqno rolling over will not cause us to stop accepting DTMF.
(closes issue 0017571)
Reported by: mdeneen
Patches:
rtp_seqno_rollover.patch uploaded by mmichelson (license 60)
Tested by: richardf, maxochoa, JJCinAZ
* Clear the AST_CDR_FLAG_DIALED flag for channels going into the pbx
via the G option in app_dial
(closes issue 0017592)
Reported by: jamicque
Patches:
G-flag-cdr-fix1.diff uploaded by mnicholson (license 96)
Tested by: jamicque, mnicholson
2010-06-29 Leif Madsen <lmadsen@digium.com>
* Release Asterisk 1.6.2.10-rc1
2010-06-28 21:51 +0000 [r272924-272927] Tilghman Lesher <tlesher@digium.com>
* main/asterisk.c, /: Merged revisions 272926 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r272926 | tilghman | 2010-06-28 16:50:57 -0500 (Mon, 28 Jun 2010)
| 15 lines Merged revisions 272925 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r272925 | tilghman | 2010-06-28 16:50:02 -0500 (Mon, 28 Jun 2010)
| 8 lines Don't change ownership/group/permissions on run
directory, if it already exists. (closes issue #17076) Reported
by: stuarth Patches: 20100324__issue17076.diff.txt uploaded by
tilghman (license 14) Tested by: stuarth ........
................
* main/config.c, /: Merged revisions 272923 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r272923 | tilghman | 2010-06-28 16:42:52 -0500 (Mon, 28 Jun 2010)
| 19 lines Merged revisions 272921-272922 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r272921 | tilghman | 2010-06-28 16:29:27 -0500 (Mon, 28 Jun 2010)
| 8 lines Change the way that we read include files, to
accommodate for changes in GCC 4.4. (closes issue #17472)
Reported by: seandarcy Patches: config2.patch uploaded by nivan
(license 1066) Tested by: nivan ........ r272922 | tilghman |
2010-06-28 16:38:49 -0500 (Mon, 28 Jun 2010) | 2 lines Also trim
trailing blanks on #includes ........ ................
2010-06-28 18:50 +0000 [r272882] Russell Bryant <russell@digium.com>
* tests/test_astobj2.c (added): Backport applicable parts of
test_astobj2 from trunk.
2010-06-28 17:37 +0000 [r272806] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 272805 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r272805 | mmichelson | 2010-06-28 12:33:12 -0500 (Mon, 28 Jun
2010) | 11 lines Merged revisions 272804 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r272804 | mmichelson | 2010-06-28 12:31:40 -0500 (Mon, 28 Jun
2010) | 5 lines Decode URI in contact header of 302 response.
ABE-2352 ........ ................
2010-06-28 15:36 +0000 [r272685-272686] Russell Bryant <russell@digium.com>
* doc/tex/chan-mobile.tex (removed): remove accidentally added
file.
* doc/tex/cdrdriver.tex, doc/tex/asterisk.tex, /,
doc/tex/chan-mobile.tex (added): Merged revisions 272684 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r272684 | russell | 2010-06-28 10:33:32 -0500 (Mon, 28
Jun 2010) | 2 lines Use the underscore package so that
underscores do not need to be escaped. ........
2010-06-25 20:20 +0000 [r272556-272577] Tilghman Lesher <tlesher@digium.com>
* /, doc/voicemail_odbc_postgresql.txt: Merged revisions 272568 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r272568 | tilghman | 2010-06-25 15:18:47 -0500
(Fri, 25 Jun 2010) | 12 lines Merged revisions 272562 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r272562 | tilghman | 2010-06-25 15:17:37 -0500 (Fri, 25 Jun 2010)
| 5 lines Make the structure of the table specified before match
the queries and results. (closes issue #17557) Reported by: cmaj
........ ................
* sounds/Makefile, /: Merged revisions 272533 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r272533 |
tilghman | 2010-06-25 14:17:16 -0500 (Fri, 25 Jun 2010) | 2 lines
Symlink sounds files, to save disk space, when multiple
tarballs/checkouts are on the same system. ........
2010-06-25 18:58 +0000 [r272531] Russell Bryant <russell@digium.com>
* include/asterisk/_private.h, tests/test_sched.c, main/asterisk.c,
include/asterisk/test.h (added), build_tools/cflags-devmode.xml,
tests/test_heap.c, tests/test_skel.c, main/Makefile, main/test.c
(added): Backport unit test API from trunk. Also, update existing
test modules that were already in this branch but had been
converted to the unit test API in trunk. Review:
https://reviewboard.asterisk.org/r/748/
2010-06-24 22:19 +0000 [r272459] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 272447 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r272447 | rmudgett | 2010-06-24 17:11:26 -0500
(Thu, 24 Jun 2010) | 17 lines Merged revisions 272446 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r272446 | rmudgett | 2010-06-24 16:58:49 -0500 (Thu, 24 Jun 2010)
| 10 lines ss_thread calls pri_grab without lock during overlap
dial Recent changes to chan_dahdi with relation to overlap
dialing call pri_grab without first obtaining a lock. (closes
issue #17414) Reported by: pdf Patches: bug17414.patch uploaded
by jpeeler (license 325) ........ ................
2010-06-23 23:40 +0000 [r272440] Terry Wilson <twilson@digium.com>
* autoconf/ast_ext_tool_check.m4, /, configure: Merged revisions
272254,272256 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r272254 |
twilson | 2010-06-23 15:53:48 -0500 (Wed, 23 Jun 2010) | 10 lines
Honor the --with-${library}=path for AST_EXT_TOOL_CHECK (closes
issue #16991) Reported by: pprindeville Patches:
with_netsnmp.patch.txt uploaded by twilson (license 396) Tested
by: twilson Review: https://reviewboard.asterisk.org/r/739/
........ r272256 | twilson | 2010-06-23 15:59:17 -0500 (Wed, 23
Jun 2010) | 2 lines Update configure when changing autconf m4
files... ........
2010-06-23 23:14 +0000 [r272371] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c, /: Merged revisions 272370 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r272370 | russell | 2010-06-23 18:09:28 -0500 (Wed, 23 Jun 2010)
| 23 lines Resolve some errors produced during module unload of
chan_iax2. The external test suite stops Asterisk using the "core
stop gracefully" command. The logs from the tests show that there
are a number of problems with Asterisk trying to cleanly shut
down. This patch addresses the following type of error that comes
from chan_iax2: [Jun 22 16:58:11] ERROR[29884]: lock.c:129
__ast_pthread_mutex_destroy: chan_iax2.c line 11371
(iax2_process_thread_cleanup): Error destroying mutex
&thread->lock: Device or resource busy For an example in the
context of a build, see:
http://bamboo.asterisk.org/browse/AST-TRUNK-739/log The primary
purpose of this patch is to change the thread pool shutdown
procedure to be more explicit to ensure that the thread exits
from a point where it is not holding a lock. While testing that,
I encountered various crashes due to the order of operations in
unload_module() being problematic. I reordered some things there,
as well. Review: https://reviewboard.asterisk.org/r/736/ ........
2010-06-23 22:37 +0000 [r272369] Matthew Nicholson <mnicholson@digium.com>
* apps/app_queue.c, /: Merged revisions 272368 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r272368 | mnicholson | 2010-06-23 17:36:49 -0500 (Wed, 23 Jun
2010) | 16 lines Merged revisions 272367 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 This version
of the patch only adds AgentComplete for attended transfers. It
was already present for blind transfers. ........ r272367 |
mnicholson | 2010-06-23 17:33:51 -0500 (Wed, 23 Jun 2010) | 8
lines Send AgentComplete manager events in the event of blind and
attended transfers. (closes issue #16819) Reported by: elbriga
Patches: app_queue.diff uploaded by elbriga (license 482)
........ ................
2010-06-23 21:54 +0000 [r272333] Tilghman Lesher <tlesher@digium.com>
* res/res_musiconhold.c, /: Merged revisions 272332 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r272332 | tilghman | 2010-06-23 16:53:49 -0500 (Wed, 23 Jun 2010)
| 8 lines If there is realtime configuration, it does not get
re-read on reload unless the config file also changes. (closes
issue #16982) Reported by: dmitri Patches: res_musiconhold.patch
uploaded by dmitri (license 1001) Tested by: atis ........
2010-06-23 21:15 +0000 [r272263] Paul Belanger <paul.belanger@polybeacon.com>
* apps/app_meetme.c: Revert previous commit, ast_test_flag64 does
not exist in 1.6.2
2010-06-23 21:09 +0000 [r272262] Tilghman Lesher <tlesher@digium.com>
* res/ael/ael.flex, /, res/ael/ael.tab.c, res/ael/ael.y,
res/ael/ael_lex.c: Merged revisions 272260 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r272260 |
tilghman | 2010-06-23 16:06:40 -0500 (Wed, 23 Jun 2010) | 8 lines
Ensure a NULL file while debugging cannot crash AEL. (closes
issue #17215) Reported by: vazir Patches:
20100518__issue17215.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman ........
2010-06-23 21:07 +0000 [r272253-272261] Paul Belanger <paul.belanger@polybeacon.com>
* /, apps/app_meetme.c: Merged revisions 272259 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r272259 |
pabelanger | 2010-06-23 17:06:15 -0400 (Wed, 23 Jun 2010) | 2
lines Fix previous merge. ast_test_flag != ast_test_flag64
........
* /, apps/app_meetme.c: Merged revisions 272257 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r272257 | pabelanger | 2010-06-23 17:00:00 -0400 (Wed, 23 Jun
2010) | 19 lines Merged revisions 272255 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r272255 | pabelanger | 2010-06-23 16:57:01 -0400 (Wed, 23 Jun
2010) | 12 lines First caller into a dynamic conference now enter
pin once. If MeetMe is configured to use dynamic conference
numbers, then the first caller (which creates the conference) had
to enter the PIN number twice. (closes issue #15878) Reported by:
shawkris Patches: issue15878.patch uploaded by pabelanger
(license 224) Tested by: pabelanger ........ ................
* main/manager.c, /: Merged revisions 272252 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r272252 |
pabelanger | 2010-06-23 16:35:45 -0400 (Wed, 23 Jun 2010) | 8
lines Correct manager variable 'EventList' case. (closes issue
#17520) Reported by: kobaz Patches: manager.patch uploaded by
kobaz (license 834) Tested by: lmadsen ........
2010-06-23 18:41 +0000 [r272124-272149] Terry Wilson <twilson@digium.com>
* /, apps/app_meetme.c: Merged revisions 272146 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r272146 |
twilson | 2010-06-23 13:39:20 -0500 (Wed, 23 Jun 2010) | 2 lines
Don't start the sla thread unless we realy need it ........
* /, apps/app_meetme.c: Merged revisions 272109 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r272109 |
twilson | 2010-06-23 12:21:40 -0500 (Wed, 23 Jun 2010) | 12 lines
Make sure reload updates SLA config Even if there are no stations
or trunks defined, we need to start the sla thread to make sure
we get the reload event. Also, when doing a reload we need to
remove the existing trunks and stations or they end up hanging
around. (closes issue #16818) Reported by: mbonin Patches:
sla_reload.patch uploaded by twilson (license 396) Tested by:
twilson ........
2010-06-22 22:14 +0000 [r272015] David Vossel <dvossel@digium.com>
* pbx/pbx_config.c, /: Merged revisions 272014 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r272014 |
dvossel | 2010-06-22 17:11:50 -0500 (Tue, 22 Jun 2010) | 5 lines
fixes issue with 'dialplan remove extension blah' segfaulting
with tab completion (closes issue #17440) Reported by: kobaz
........
2010-06-22 17:37 +0000 [r271904] Matthew Nicholson <mnicholson@digium.com>
* /, channels/chan_sip.c: Merged revisions 271903 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r271903 | mnicholson | 2010-06-22 12:35:17 -0500 (Tue, 22 Jun
2010) | 15 lines Merged revisions 271902 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r271902 | mnicholson | 2010-06-22 12:31:57 -0500 (Tue, 22 Jun
2010) | 8 lines Decrease the module ref count in sip_hangup when
SIP_DEFER_BYE_ON_TRANSFER is set. This is necessary to keep the
ref count correct. (closes issue #16815) Reported by: rain
Patches: chan_sip-unref-fix.diff uploaded by rain (license 327)
(modified) Tested by: rain ........ ................
2010-06-22 16:30 +0000 [r271869] Russell Bryant <russell@digium.com>
* /, res/ais/clm.c, res/ais/evt.c: Merged revisions 271867 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r271867 | russell | 2010-06-22 11:28:03 -0500 (Tue, 22
Jun 2010) | 7 lines Resolve some errors that occur on a graceful
shutdown. Don't Finalize() if Initialize() did not succeed. This
resulted in an error about trying to Finalize() an invalid
handle. Also trim some trailing whitespace while in the area.
........
2010-06-22 15:49 +0000 [r271832] David Vossel <dvossel@digium.com>
* /, main/features.c: Merged revisions 271831 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r271831 |
dvossel | 2010-06-22 10:46:22 -0500 (Tue, 22 Jun 2010) | 10 lines
fixes attended transfer behavior when both transferee and
transferer hung up If both the transferer and transferee of a
attended transfer hangup before the new channel picks up, the new
channel should be hung up as well as it has no endpoint to talk
to. This mirrors the expected behavior used in 1.4. (closes issue
#17444) Reported by: corruptor ........
2010-06-22 15:00 +0000 [r271691-271763] Matthew Nicholson <mnicholson@digium.com>
* configs/dundi.conf.sample, /, pbx/pbx_dundi.c: Merged revisions
271762 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r271762 | mnicholson | 2010-06-22 09:54:58 -0500 (Tue, 22 Jun
2010) | 15 lines Merged revisions 271761 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r271761 | mnicholson | 2010-06-22 09:49:36 -0500 (Tue, 22 Jun
2010) | 9 lines Allow users to specify a port for dundi peers.
(closes issue #17056) Reported by: klaus3000 Patches:
dundi-peerport-patch-trunk.txt uploaded by klaus3000 (license 65)
Tested by: klaus3000 ........ ................
* include/asterisk/strings.h, configs/sip_notify.conf.sample, /,
channels/chan_sip.c: Merged revisions 271690 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r271690 | mnicholson | 2010-06-22 07:58:28 -0500 (Tue, 22 Jun
2010) | 18 lines Merged revisions 271689 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r271689 | mnicholson | 2010-06-22 07:52:27 -0500 (Tue, 22 Jun
2010) | 8 lines Modify chan_sip's packet generation api to
automatically calculate the Content-Length. This is done by
storing packet content in a buffer until it is actually time to
send the packet, at which time the size of the packet is
calculated. This change was made to ensure that the
Content-Length is always correct. (closes issue #17326) Reported
by: kenner Tested by: mnicholson, kenner Review:
https://reviewboard.asterisk.org/r/693/ ........ This change also
adds an ast_str_copy_string() function (similar to
ast_copy_string), that copies one ast_str into another, properly
handling embedded nulls. ................
2010-06-21 20:48 +0000 [r271555] Jeff Peeler <jpeeler@digium.com>
* res/ael/pval.c, /: Merged revisions 271554 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r271554 | jpeeler | 2010-06-21 15:46:53 -0500 (Mon, 21 Jun 2010)
| 14 lines Merged revisions 271552 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r271552 | jpeeler | 2010-06-21 15:37:47 -0500 (Mon, 21 Jun 2010)
| 7 lines Do not use sizeof to calculate size of a heap allocated
character array. Change left out from 271399. (closes issue
#16053) Reported by: diLLec ........ ................
2010-06-18 21:33 +0000 [r271338-271484] Jeff Peeler <jpeeler@digium.com>
* res/ael/pval.c, /, include/asterisk/pval.h, pbx/pbx_ael.c: Merged
revisions 271483 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r271483 | jpeeler | 2010-06-18 16:32:09 -0500 (Fri, 18 Jun 2010)
| 18 lines Merged revisions 271399 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r271399 | jpeeler | 2010-06-18 14:28:24 -0500 (Fri, 18 Jun 2010)
| 11 lines Fix crash when parsing some heavily nested statements
in AEL on reload. Due to the recursion used when compiling AEL in
gen_prios, all the stack space was being consumed when parsing
some AEL that contained nesting 13 levels deep. Changing a few
large buffers to be heap allocated fixed the crash, although I
did not test how many more levels can now be safely used. (closes
issue #16053) Reported by: diLLec Tested by: jpeeler ........
................
* channels/chan_dahdi.c, /: Merged revisions 269307 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r269307 | rmudgett | 2010-06-09 11:54:38 -0500 (Wed, 09 Jun 2010)
| 12 lines Eliminate deadlock potential in dahdi_fixup(). Calling
dahdi_indicate() within dahdi_fixup() while the owner pointers
are in a potentially inconsistent state is a potentially bad
thing in principle. However, calling dahdi_indicate() when the
channel private lock is already held can cause a deadlock if the
PRI lock is needed because dahdi_indicate() will also get the
channel private lock. The pri_grab() function assumes that the
channel private lock is held once to avoid deadlock. ........
2010-06-17 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.9 Released.
2010-06-10 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.9-rc3 Released.
2010-06-10 Tilghman Lesher <tlesher@digium.com>
* Ensure signals are not blocked inside other signal handlers.
This eliminates the annoying <beep> on the console.
(closes issue 0017477)
Reported by: jvandal
Patches:
20100610__issue17477.diff.txt uploaded by tilghman (license 14
2010-06-09 Paul Belanger <paul.belanger@polybeacon.com>
* Fix Debian init script to not use -c.
When using the init script as-is currently, it could cause issues on Debian
such as high CPU usage. This fix has worked for several people so I'm
implementing the change. We now handle color displays properly.
(closes issue 0016784)
Reported by: pabelanger
Patches:
20100530__issue16784__2.diff.txt uploaded by tilghman (license 14)
Tested by: pabelanger, tilghman
2010-06-07 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.9-rc2 Released.
2010-06-07 Tilghman Lesher <tlesher@digium.com>
* Fix crash in DTMF detection.
What I did not originally see in my previous commit was that even
though the next digit could be detected before the previous was
considered ended, the detection of the next digit effectively ends
the detection of the previous. Therefore, the length moves in
lockstep with the digit, and no separate counter is needed for the
length alone.
(closes issue 0017371)
Reported by: alecdavis
(closes issue 0017474)
Reported by: kenner
2010-06-01 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.9-rc1 Released.
2010-06-01 15:20 +0000 [r266598] Tilghman Lesher <tlesher@digium.com>
* main/asterisk.c, /: Merged revisions 266592 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r266592 | tilghman | 2010-06-01 10:18:59 -0500 (Tue, 01 Jun 2010)
| 18 lines Merged revisions 266585 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r266585 | tilghman | 2010-06-01 10:17:46 -0500 (Tue, 01 Jun 2010)
| 11 lines Prevent CLI prompt from distorting output of lines
shorter than the prompt. Uses the VT100 method of clearing the
line from the cursor position to the end of the line: Esc-0K
(closes issue #17160) Reported by: coolmig Patches:
20100531__issue17160.diff.txt uploaded by tilghman (license 14)
Tested by: coolmig ........ ................
2010-05-31 16:07 +0000 [r266570] Paul Belanger <paul.belanger@polybeacon.com>
* res/res_agi.c: Fix typo in documentation (closes issue #17395)
Reported by: pabelanger Patches: res_agi.c.patch uploaded by
pabelanger (license 224)
2010-05-30 04:45 +0000 [r266439] Tilghman Lesher <tlesher@digium.com>
* contrib/init.d/rc.debian.asterisk, /: Merged revisions 266438 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r266438 | tilghman | 2010-05-29 23:44:28 -0500
(Sat, 29 May 2010) | 9 lines Merged revisions 266437 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r266437 | tilghman | 2010-05-29 23:43:28 -0500 (Sat, 29
May 2010) | 2 lines Reverting patch and reopening issue #16784,
as patch breaks color display. ........ ................
2010-05-28 20:55 +0000 [r266338] Tilghman Lesher <tlesher@digium.com>
* main/asterisk.c, /: Merged revisions 266337 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r266337 |
tilghman | 2010-05-28 15:53:04 -0500 (Fri, 28 May 2010) | 1 line
Only report swap on platforms which can examine those statistics
........
2010-05-28 17:57 +0000 [r266293] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 266292 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r266292 |
dvossel | 2010-05-28 12:55:38 -0500 (Fri, 28 May 2010) | 9 lines
fixes crash when creation of UDPTL fails (closes issue #17264)
Reported by: falves11 Patches: issue_17264_reviewboard_fix.diff
uploaded by dvossel (license 671)
issue_17264_1.6.2_reviewboard_fix.diff uploaded by dvossel
(license 671) Tested by: falves11 ........
2010-05-26 21:19 +0000 [r266154] Tilghman Lesher <tlesher@digium.com>
* utils/extconf.c, main/asterisk.c, /, main/logger.c: Merged
revisions 266146 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r266146 | tilghman | 2010-05-26 16:17:46 -0500 (Wed, 26 May 2010)
| 21 lines Merged revisions 266142 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r266142 | tilghman | 2010-05-26 16:11:44 -0500 (Wed, 26 May 2010)
| 14 lines Use sigaction for signals which should persist past
the initial trigger, not signal. If you call signal() in a
Solaris signal handler, instead of just resetting the signal
handler, it causes the signal to refire, because the signal is
not marked as handled prior to the signal handler being called.
This effectively causes Solaris to immediately exceed the
threadstack in recursive signal handlers and crash. (closes issue
#17000) Reported by: rmcgilvr Patches:
20100526__issue17000.diff.txt uploaded by tilghman (license 14)
Tested by: rmcgilvr ........ ................
2010-05-26 18:37 +0000 [r266007] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 266006 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r266006 |
dvossel | 2010-05-26 13:32:51 -0500 (Wed, 26 May 2010) | 8 lines
fixes failed SIP Directed pickup resulting in dead channel
(closes issue #17339) Reported by: one47 Patches:
sip_magic_pickup2 uploaded by one47 (license 23) Tested by:
one47, dvossel ........
2010-05-26 16:31 +0000 [r265895-265959] Tilghman Lesher <tlesher@digium.com>
* res/res_config_pgsql.c, /: Merged revisions 265923 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r265923 | tilghman | 2010-05-26 11:23:28 -0500
(Wed, 26 May 2010) | 14 lines Merged revisions 265910 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r265910 | tilghman | 2010-05-26 11:21:00 -0500 (Wed, 26 May 2010)
| 7 lines Not finding rows in the DB does not rise to the level
of a warning. (closes issue #17062) Reported by: drookie Patches:
20100525__issue17062.diff.txt uploaded by tilghman (license 14)
........ ................
* configs/res_pgsql.conf.sample, res/res_config_pgsql.c, /: Merged
revisions 265894 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r265894 |
tilghman | 2010-05-26 11:14:48 -0500 (Wed, 26 May 2010) | 8 lines
Construct socket name, according to the Postgres docs, and
document as such. (closes issue #17392) Reported by: dps Patches:
20100525__issue17392.diff.txt uploaded by tilghman (license 14)
Tested by: dps ........
2010-05-26 15:52 +0000 [r265890] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Recorded merge of revisions 265842 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r265842 | mmichelson | 2010-05-26 09:41:55 -0500 (Wed,
26 May 2010) | 9 lines Re-enable "always" option for videosupport
option in sip.conf. (closes issue #17016) Reported by: twilson
Patches: 17016.patch uploaded by mmichelson (license 60) Tested
by: devmod ........
2010-05-26 00:33 +0000 [r265748] Tilghman Lesher <tlesher@digium.com>
* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
pbx/pbx_lua.c: Merged revisions 265747 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r265747 |
tilghman | 2010-05-25 19:29:40 -0500 (Tue, 25 May 2010) | 8 lines
Use configure to determine the prefixes and include directories
properly. This ensures cross-platform compatibility, even among
Linux distributions, which don't always put headers in the same
place. (closes issue #17391) Reported by: loloski ........
2010-05-25 21:05 +0000 [r265699] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 265698 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r265698 |
mmichelson | 2010-05-25 15:59:04 -0500 (Tue, 25 May 2010) | 12
lines Properly use peer's outboundproxy for outbound REGISTERs.
The logic used in transmit_register to get the outboundproxy for
a peer was flawed since this value would be overridden shortly
afterwards when create_addr was called. In addition, this also
fixes some logic used when parsing users.conf so that the peer
name is placed in the internally-generated register string so
that an outboundproxy set in the Asterisk GUI will be used for
outbound REGISTERs. ........
2010-05-25 17:15 +0000 [r265615] David Vossel <dvossel@digium.com>
* channels/chan_dahdi.c: fixes build issue with zaptel (closes
issue 0017394) Reported by: aragon Patches: half_buffer_fix.diff
uploaded by dvossel (license 671) Tested by: aragon
2010-05-25 17:06 +0000 [r265612] Matthew Nicholson <mnicholson@digium.com>
* apps/app_queue.c, /: Merged revisions 265611 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r265611 | mnicholson | 2010-05-25 12:00:11 -0500 (Tue, 25 May
2010) | 15 lines Merged revisions 265610 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r265610 | mnicholson | 2010-05-25 11:48:19 -0500 (Tue, 25 May
2010) | 8 lines Don't mark the cdr records of unanswered queue
calls with "NOANSWER". This restores the behavior prior to
r258670. (closes issue #17334) Reported by: jvandal Patches:
queue-cdr-fixes1.diff uploaded by mnicholson (license 96) Tested
by: aragon, jvandal ........ ................
2010-05-24 23:52 +0000 [r265521] Terry Wilson <twilson@digium.com>
* include/asterisk/options.h, main/asterisk.c, Makefile,
doc/manager_1_1.txt, doc/tex/manager.tex, main/manager.c: Merged
revisions 265320,265467 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r265320 |
twilson | 2010-05-24 14:06:40 -0500 (Mon, 24 May 2010) | 14 lines
Add the FullyBooted AMI event It is possible to connect to the
manager interface before all Asterisk modules are loaded. To
ensure that an application does not send AMI actions that might
require a module that has not yet loaded, the application can
listen for the FullyBooted manager event. It will be sent upon
connection if all modules have been loaded, or as soon as loading
is complete. The event: Event: FullyBooted Privilege: system,all
Status: Fully Booted Review:
https://reviewboard.asterisk.org/r/639/ ........ r265467 |
twilson | 2010-05-24 17:21:58 -0500 (Mon, 24 May 2010) | 1 line
Merge the rest of the FullyBooted patch ........
2010-05-24 22:07 +0000 [r265450-265452] Mark Michelson <mmichelson@digium.com>
* /, channels/h323/ast_h323.cxx: Merged revisions 265451 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r265451 | mmichelson | 2010-05-24 17:05:15 -0500 (Mon,
24 May 2010) | 8 lines Print openh323 log to the Asterisk
console. (closes issue #17109) Reported by: under Patches:
logstream.diff uploaded by under (license 914) ........
* /, channels/chan_sip.c: Merged revisions 265449 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r265449 |
mmichelson | 2010-05-24 16:44:30 -0500 (Mon, 24 May 2010) | 11
lines Allow type=user SIP endpoints to be loaded properly from
realtime. (closes issue #16021) Reported by: Guggemand Patches:
realtime-type-fix.patch uploaded by Guggemand (license 897)
(altered by me slightly to avoid ref leaks) Tested by: Guggemand
........
2010-05-24 19:30 +0000 [r265364] David Vossel <dvossel@digium.com>
* main/channel.c, /: Merged revisions 265273 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r265273 |
dvossel | 2010-05-24 11:10:09 -0500 (Mon, 24 May 2010) | 2 lines
fixes segfault when using generic plc ........
2010-05-24 18:30 +0000 [r265318] Tilghman Lesher <tlesher@digium.com>
* main/asterisk.c, /: Merged revisions 265316 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r265316 |
tilghman | 2010-05-24 13:19:08 -0500 (Mon, 24 May 2010) | 7 lines
On systems with a LOT of RAM, a signed integer sometimes printed
negative. (closes issue #16837) Reported by: jlpedrosa Patches:
20100504__issue16837.diff.txt uploaded by tilghman (license 14)
........
2010-05-21 21:57 +0000 [r264998-265172] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Fix memory hogging behavior of app_queue. From
reviewboard: This review request is for the patch on issue 17081.
A user reported that he saw increasing numbers of allocations
stemming from app_queue.c when he would run the "queue show" CLI
command. The user reported that he was using approximately 40
realtime queues and as he ran the CLI command more and more, the
memory usage would shoot up. As it turns out, there was a memory
leak and a separate usage of memory that, while not really a
leak, was very irresponsible. Both memory problems can be
attributed to the function init_queue(). When the "queue show"
command is run, all realtime queues have the init_queue()
function called on the in-memory queue. The idea is to place the
queue in its default state and then overwrite options specified
in the realtime backend as we read them. The first problem, the
memory leak, had to do with the fact that the string field for
the name of the first periodic announcement file was being
re-created every time init_queue was called. This patch corrects
the behavior by only calling ast_str_create if the memory has not
already been allocated. The other problem is a bit more
complicated. The majority of the strings in the call_queue
structure were changed to use the ast_string_fields API for 1.6.0
and beyond. init_queue resets all string fields on the queue to
their default values. Then, later in the realtime queue loading
process, these string fields are set to their configured values.
For those unfamiliar with string fields, frequent resizing of a
string like this is not what the string fields API is designed
for. The result of this constant resizing is that as the queue
gets loaded, eventually space for the string runs out and so a
new memory pool, at twice the size of the previously allocated
one, is created for the string fields. The reporter of issue
17081 wrote a script that ran the "queue show" CLI command 2100
times. By the end, each of his 40 queues was taking about a
megabyte of memory apiece just for their string fields. My fix
for this problem is to revert the call_queue structure from using
string fields. In my patch here, I have moved the queue back to
using fixed-sized buffers. I ran the script provided by the
reporter of 17081 and determined that I no longer saw the
steadily-increasing memory usage that I had seen before applying
the patch. (closes issue #17081) Reported by: wliegel Patches:
17081v2.patch uploaded by mmichelson (license 60) Tested by:
wliegel, mmichelson Review:
https://reviewboard.asterisk.org/r/651/
* apps/app_queue.c, include/asterisk/file.h, /: Merged revisions
265090 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r265090 | mmichelson | 2010-05-21 16:08:51 -0500 (Fri, 21 May
2010) | 15 lines Merged revisions 265089 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r265089 | mmichelson | 2010-05-21 15:59:14 -0500 (Fri, 21 May
2010) | 8 lines Don't hang up on a queue caller if the file we
attempt to play does not exist. This also fixes a documentation
mistake in file.h that made my original attempt to correct this
problem not work correctly. (closes issue #17061) Reported by:
RoadKill ........ ................
* /, channels/chan_sip.c: Merged revisions 265087 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r265087 |
mmichelson | 2010-05-21 15:38:14 -0500 (Fri, 21 May 2010) | 7
lines Be sure to set the sin_family on the proxy when allocating.
(closes issue #17157) Reported by: stuarth ........
* /, include/asterisk/channel.h: Merged revisions 265000 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r265000 | mmichelson | 2010-05-21 11:54:21 -0500
(Fri, 21 May 2010) | 9 lines Merged revisions 264999 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r264999 | mmichelson | 2010-05-21 11:53:53 -0500 (Fri,
21 May 2010) | 3 lines Fix grammatical error in comment. ........
................
* main/channel.c, main/autoservice.c, /,
include/asterisk/channel.h: Merged revisions 264997 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r264997 | mmichelson | 2010-05-21 11:44:27 -0500
(Fri, 21 May 2010) | 38 lines Merged revisions 264996 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r264996 | mmichelson | 2010-05-21 11:28:34 -0500 (Fri, 21 May
2010) | 32 lines Allow ast_safe_sleep to defer specific frames
until after the sleep has concluded. From reviewboard Background:
A Digium customer discovered a somewhat odd bug. The setup is
that parties A and B are bridged, and party A places party B on
hold. While party B is listening to hold music, he mashes a bunch
of DTMF. Party A takes party B off hold while this is happening,
but party B continues to hear hold music. I could reproduce this
about 1 in 5 times. The issue: When DTMF features are enabled and
a user presses keys, the channel that the DTMF is streamed to is
placed in an ast_safe_sleep for 100 ms, the duration of the
emulated tone. If an AST_CONTROL_UNHOLD frame is read from the
channel during the sleep, the frame is dropped. Thus the unhold
indication is never made to the channel that was originally
placed on hold. The fix: Originally, I discussed with Kevin
possible ways of fixing the specific problem reported. However,
we determined that the same type of problem could happen in other
situations where ast_safe_sleep() is used. Using autoservice as a
model, I modified ast_safe_sleep_conditional() to defer specific
frame types so they can be re-queued once the sleep has finished.
I made a common function for determining if a frame should be
deferred so that there are not two identical switch blocks to
maintain. Review: https://reviewboard.asterisk.org/r/674/
........ ................
2010-05-20 23:34 +0000 [r264829] Richard Mudgett <rmudgett@digium.com>
* /, main/callerid.c: Merged revisions 264828 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r264828 | rmudgett | 2010-05-20 18:29:43 -0500 (Thu, 20 May 2010)
| 13 lines Merged revisions 264820 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r264820 | rmudgett | 2010-05-20 18:23:21 -0500 (Thu, 20 May 2010)
| 6 lines ast_callerid_parse() had a path that left name
uninitialized. Several callers of ast_callerid_parse() do not
initialize the name parameter before calling thus there is the
potential to use an uninitialized pointer. ........
................
2010-05-20 22:24 +0000 [r264753-264783] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c, /: Merged revisions 264779 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r264779 |
tilghman | 2010-05-20 17:23:32 -0500 (Thu, 20 May 2010) | 8 lines
Let ExtensionState resolve dynamic hints. (closes issue #16623)
Reported by: tilghman Patches: 20100116__issue16623.diff.txt
uploaded by tilghman (license 14) Tested by: lmadsen ........
* apps/app_stack.c, /: Merged revisions 264752 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r264752 |
tilghman | 2010-05-20 16:28:53 -0500 (Thu, 20 May 2010) | 7 lines
Error message fix. (closes issue #17356) Reported by: kenner
Patches: app_stack.c.diff uploaded by kenner (license 1040)
........
2010-05-19 22:10 +0000 [r264453] Mark Michelson <mmichelson@digium.com>
* include/asterisk/_private.h, include/asterisk/options.h,
main/asterisk.c, main/loader.c, main/channel.c, /,
channels/chan_sip.c: Merged revisions 264452 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r264452 |
mmichelson | 2010-05-19 16:29:08 -0500 (Wed, 19 May 2010) | 86
lines Fix transcode_via_sln option with SIP calls and improve PLC
usage. From reviewboard: The problem here is a bit complex, so
try to bear with me... It was noticed by a Digium customer that
generic PLC (as configured in codecs.conf) did not appear to
actually be having any sort of benefit when packet loss was
introduced on an RTP stream. I reproduced this issue myself by
streaming a file across an RTP stream and dropping approx. 5% of
the RTP packets. I saw no real difference between when PLC was
enabled or disabled when using wireshark to analyze the RTP
streams. After analyzing what was going on, it became clear that
one of the problems faced was that when running my tests, the
translation paths were being set up in such a way that PLC could
not possibly work as expected. To illustrate, if packets are lost
on channel A's read stream, then we expect that PLC will be
applied to channel B's write stream. The problem is that generic
PLC can only be done when there is a translation path that moves
from some codec to SLINEAR. When I would run my tests, I found
that every single time, read and write translation paths would be
set up on channel A instead of channel B. There appeared to be no
real way to predict which channel the translation paths would be
set up on. This is where Kevin swooped in to let me know about
the transcode_via_sln option in asterisk.conf. It is supposed to
work by placing a read translation path on both channels from the
channel's rawreadformat to SLINEAR. It also will place a write
translation path on both channels from SLINEAR to the channel's
rawwriteformat. Using this option allows one to predictably set
up translation paths on all channels. There are two problems with
this, though. First and foremost, the transcode_via_sln option
did not appear to be working properly when I was placing a SIP
call between two endpoints which did not share any common
formats. Second, even if this option were to work, for PLC to be
applied, there had to be a write translation path that would go
from some format to SLINEAR. It would not work properly if the
starting format of translation was SLINEAR. The one-line change
presented in this review request in chan_sip.c fixed the first
issue for me. The problem was that in sip_request_call, the
jointcapability of the outbound channel was being set to the
format passed to sip_request_call. This is nativeformats of the
inbound channel. Because of this, when
ast_channel_make_compatible was called by app_dial, both channels
already had compatibly read and write formats. Thus, no
translation path was set up at the time. My change is to set the
jointcapability of the sip_pvt created during sip_request_call to
the intersection of the inbound channel's nativeformats and the
configured peer capability that we determined during the earlier
call to create_addr. Doing this got the translation paths set up
as expected when using transcode_via_sln. The changes presented
in channel.c fixed the second issue for me. First and foremost,
when Asterisk is started, we'll read codecs.conf to see the value
of the genericplc option. If this option is set, and ast_write is
called for a frame with no data, then we will attempt to fill in
the missing samples for the frame. The implementation uses a
channel datastore for maintaining the PLC state and for creating
a buffer to store PLC samples in. Even when we receive a frame
with data, we'll call plc_rx so that the PLC state will have
knowledge of the previous voice frame, which it can use as a
basis for when it comes time to actually do a PLC fill-in. So,
reviewers, now I ask for your help. First off, there's the one
line change in chan_sip that I have put in. Is it right? By my
logic it seems correct, but I'm sure someone can tell me why it
is not going to work. This is probably the change I'm least
concerned about, though. What concerns me much more is the set of
changes in channel.c. First off, am I even doing it right? When I
run tests, I can clearly see that when PLC is activated, I see a
significant increase in RTP traffic where I would expect it to
be. However, in my humble opinion, the audio sounds kind of
crappy whenever the PLC fill-in is done. It sounds worse to me
than when no PLC is used at all. I need someone to review the
logic I have used to be sure that I'm not misusing anything. As
far as I can see my pointer arithmetic is correct, and my use of
AST_FRIENDLY_OFFSET should be correct as well, but I'm sure
someone can point out somewhere where I've done something
incorrectly. As I was writing this review request up, I decided
to give the code a test run under valgrind, and I find that for
some reason, calls to plc_rx are causing some invalid reads.
Apparently I'm reading past the end of a buffer somehow. I'll
have to dig around a bit to see why that is the case. If it's
obvious to someone reviewing, speak up! Finally, I have one other
proposal that is not reflected in my code review. Since without
transcode_via_sln set, one cannot predict or control where a
translation path will be up, it seems to me that the current
practice of using PLC only when transcoding to SLINEAR is not
useful. I recommend that once it has been determined that the
method used in this code review is correct and works as expected,
then the code in translate.c that invokes PLC should be removed.
Review: https://reviewboard.asterisk.org/r/622/ ........
2010-05-19 20:31 +0000 [r264405] David Vossel <dvossel@digium.com>
* main/udptl.c, /: Merged revisions 264400 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r264400 |
dvossel | 2010-05-19 15:30:33 -0500 (Wed, 19 May 2010) | 11 lines
fixes infinite loop during udptl.c's decode_open_type When
decode_length returns the length there is a check to see if that
length is negative, if so the decode loop breaks as this means
the limit has been reached. The problem here is that length is an
unsigned int, so length can never be negative. This resulted in
an infinite loop. (issue #17352) ........
2010-05-19 20:27 +0000 [r264336-264388] Matthew Nicholson <mnicholson@digium.com>
* main/udptl.c, /: Merged revisions 264379 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r264379 |
mnicholson | 2010-05-19 15:26:27 -0500 (Wed, 19 May 2010) | 4
lines Cast an unsigned int to a signed int when comparing it with
0. (AST-377) ........
* apps/app_speech_utils.c, /: Merged revisions 264335 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r264335 | mnicholson | 2010-05-19 15:02:57 -0500
(Wed, 19 May 2010) | 12 lines Merged revisions 264334 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r264334 | mnicholson | 2010-05-19 15:01:38 -0500 (Wed, 19 May
2010) | 5 lines Set quieted flag when receiving a dtmf tone
during playback in speechbackground. (closes issue #16966)
Reported by: asackheim ........ ................
2010-05-19 19:25 +0000 [r264332] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 264331 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r264331 |
dvossel | 2010-05-19 14:21:04 -0500 (Wed, 19 May 2010) | 13 lines
fixes crash in check_rtp_timeout During deadlock avoidance the
sip dialog pvt is locked and unlocked. When this occurs we have
no guarantee the pvt's owner is still valid. We were trying to
access the pvt's owner after this without checking to see if it
still existed first. (closes issue #17271) Reported by: under
Patches: check_rtp_timeout.diff uploaded by under (license 914)
Tested by: dvossel ........
2010-05-19 17:49 +0000 [r264205-264250] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/options.h, /, configure,
include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
264249 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r264249 | tilghman | 2010-05-19 12:48:31 -0500 (Wed, 19 May 2010)
| 24 lines Merged revisions 264248 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r264248 | tilghman | 2010-05-19 12:41:29 -0500 (Wed, 19 May 2010)
| 17 lines Internal timing is now on by default, if you're using
DAHDI 2.3 or above. The reason for ensuring DAHDI 2.3 or above is
that this version ensures that a timer is always available,
whereas in previous versions, it was possible for DAHDI to be
loaded, but have no drivers to actually generate timing. If
internal_timing was turned on in this circumstance, a complete
lack of audio would result. This is the reason why
internal_timing was not on by default. However, now that DAHDI
ensures the availability of a timer, there is no reason for this
setting to be off (and in fact, it solves a great many initial
user problems). (closes issue #15932) Reported by: dimas Patches:
20100519__issue15932.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman ........ ................
* main/dsp.c, /: Merged revisions 264204 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r264204 |
tilghman | 2010-05-19 11:42:20 -0500 (Wed, 19 May 2010) | 9 lines
Keep track of digit duration, when we're decoding inband to pass
DTMF frames. (closes issue #17235) Reported by: frawd Patches:
new_dtmf_dsp_len.patch uploaded by frawd (license 610)
20100518__issue17235.diff.txt uploaded by tilghman (license 14)
Tested by: frawd ........
2010-05-19 14:47 +0000 [r264115] David Vossel <dvossel@digium.com>
* main/rtp.c, /: Merged revisions 264114 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r264114 |
dvossel | 2010-05-19 09:38:02 -0500 (Wed, 19 May 2010) | 13 lines
fixes crash during dtmf During the processing of Cisco dtmf the
dtmf samples were not being calculated correctly. In an attempt
to determine what sample rate was being used, a NULL frame was
processed which caused a crash. This patch resolves this. (closes
issue #17248) Reported by: falves11 Patches: issue_17248.diff
uploaded by dvossel (license 671) ........
2010-05-19 08:15 +0000 [r264032] Alec L Davis <sivad.a@paradise.net.nz>
* /, configs/indications.conf.sample: Merged revisions 264031 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r264031 | alecdavis | 2010-05-19 20:09:14 +1200 (Wed, 19
May 2010) | 8 lines fix incorrectly typed indications for [nz]
stutter and dialrecall (closes issue #17359) Reported by:
alecdavis Patches: bug17359.diff.txt uploaded by alecdavis
(license 585) ........
2010-05-19 06:41 +0000 [r263951] Tilghman Lesher <tlesher@digium.com>
* main/dsp.c, /: Merged revisions 263950 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r263950 | tilghman | 2010-05-19 01:41:04 -0500 (Wed, 19 May 2010)
| 15 lines Merged revisions 263949 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r263949 | tilghman | 2010-05-19 01:32:27 -0500 (Wed, 19 May 2010)
| 8 lines Because progress is called multiple times, across
several frames, we must persist states when detecting multitone
sequences. (closes issue #16749) Reported by: dant Patches:
dsp.c-bug16749-1.patch uploaded by dant (license 670) Tested by:
dant ........ ................
2010-05-18 22:49 +0000 [r263906] David Vossel <dvossel@digium.com>
* main/strings.c, /: Merged revisions 263904 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r263904 |
dvossel | 2010-05-18 17:48:51 -0500 (Tue, 18 May 2010) | 9 lines
fixes segfault on logging (closes issue #17331) Reported by:
under Patches: utils.diff uploaded by under (license 914)
segfault_on_logging.diff uploaded by dvossel (license 671) Tested
by: under, dvossel ........
2010-05-18 19:41 +0000 [r263809] Jeff Peeler <jpeeler@digium.com>
* apps/app_directory.c, /: Merged revisions 263807 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r263807 | jpeeler | 2010-05-18 14:27:34 -0500
(Tue, 18 May 2010) | 17 lines Merged revisions 263769 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r263769 | jpeeler | 2010-05-18 13:54:58 -0500 (Tue, 18 May 2010)
| 10 lines Modify directory name reading to be interrupted with
operator or pound escape. In the case of accidentally entering
the wrong first three letters for the reading, users could be
very frustrated if the name listing is very long. This allows
interrupting the reading by pressing 0 or #. 0 will attempt to
execute a configured operator (o) extension and # will exit and
proceed in the dialplan. ABE-2200 ........ ................
2010-05-17 22:10 +0000 [r263642] Mark Michelson <mmichelson@digium.com>
* /, main/devicestate.c: Merged revisions 263640 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r263640 | mmichelson | 2010-05-17 17:08:01 -0500 (Mon, 17 May
2010) | 16 lines Merged revisions 263639 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r263639 | mmichelson | 2010-05-17 17:00:28 -0500 (Mon, 17 May
2010) | 10 lines Fix logic error when checking for a devstate
provider. When using strsep, if one of the list of specified
separators is not found, it is the first parameter to strsep
which is now NULL, not the pointer returned by strsep. This issue
isn't especially severe in that the worst it is likely to do is
waste some cycles when a device with no '/' and no ':' is passed
to ast_device_state. ........ ................
2010-05-17 19:37 +0000 [r263587-263590] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c, /: Merged revisions 263589 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r263589 | tilghman | 2010-05-17 14:31:15 -0500 (Mon, 17 May 2010)
| 9 lines With IMAP backend, messages in INBOX were counted twice
for MWI. (closes issue #17135) Reported by: edhorton Patches:
20100513__issue17135.diff.txt uploaded by tilghman (license 14)
17135_2.diff uploaded by ebroad (license 878) Tested by:
edhorton, ebroad ........
* main/app.c: Don't close 'n', just close 'above_n'. (closes issue
#17345) Reported by: wdoekes
2010-05-17 14:41 +0000 [r263376-263458] Leif Madsen <lmadsen@digium.com>
* main/manager.c, /: Merged revisions 263457 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r263457 | lmadsen | 2010-05-17 09:37:35 -0500 (Mon, 17 May 2010)
| 19 lines Recorded merge of revisions 263456 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r263456 | lmadsen | 2010-05-17 09:35:18 -0500 (Mon, 17 May 2010)
| 11 lines Manager cookies are not compatible with RFC2109. The
Version field in the cookies we're setting contain quotes around
the version number which is not compatible with RFC2109 and
breaks some implementations. (closes issue #17231) Reported by:
ecarruda Patches: manager_rfc2109-trunk-v1.patch uploaded by
ecarruda (license 559) manager_rfc2109-1.6.2-v1.patch uploaded by
ecarruda (license 559) Tested by: ecarruda, russell ........
................
* sounds/Makefile, /: Merged revisions 263375 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r263375 | lmadsen | 2010-05-17 09:05:33 -0500 (Mon, 17 May 2010)
| 16 lines Merged revisions 263374 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r263374 | lmadsen | 2010-05-17 09:04:57 -0500 (Mon, 17 May 2010)
| 8 lines Update link to new version of core sounds. The latest
version of the core sounds files 1.4.19 now includes the missing
queue-minute sound file which is called by app_queue but which
has been missing. (closes issue #17123) Reported by: n8ideas
........ ................
2010-05-17 13:03 +0000 [r263293] David Vossel <dvossel@digium.com>
* CHANGES, channels/chan_dahdi.c: backport of DAHDI dynamic buffer
policy dialstring option
2010-05-15 23:41 +0000 [r263202] Paul Belanger <paul.belanger@polybeacon.com>
* /, codecs/gsm/Makefile: Merged revisions 252488 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r252488 |
tilghman | 2010-03-15 12:27:08 -0400 (Mon, 15 Mar 2010) | 9 lines
Make the Makefile logic more explicit and move the Snow Leopard
logic down to where it's not executed on non-Darwin systems.
(closes issue #17028) Reported by: pabelanger Patches:
issue17028_20100315.patch uploaded by seanbright (license 71)
20100315__issue17028.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman, pabelanger ........
2010-05-13 22:13 +0000 [r263070] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 263069 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r263069 | rmudgett | 2010-05-13 17:01:36 -0500 (Thu, 13 May 2010)
| 1 line Fix inverted logic in cli command: ss7 set debug on/off
........
2010-05-13 15:36 +0000 [r262898] Russell Bryant <russell@digium.com>
* channels/chan_console.c, /: Merged revisions 262897 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r262897 | russell | 2010-05-13 10:36:12 -0500 (Thu, 13 May 2010)
| 4 lines Fix an off by one error that causes a crash. Thanks to
Raymond Burke for pointing it out. ........
2010-05-12 20:01 +0000 [r262801] Paul Belanger <paul.belanger@polybeacon.com>
* main/loader.c, main/cli.c, /: Merged revisions 262800 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r262800 | pabelanger | 2010-05-12 15:59:16 -0400 (Wed,
12 May 2010) | 8 lines Notify CLI when modules is loaded /
unloaded (closes issue #17308) Reported by: pabelanger Patches:
cli.modules.patch uploaded by pabelanger (license 224) Tested by:
pabelanger, russell ........
2010-05-12 19:53 +0000 [r262797-262799] Leif Madsen <lmadsen@digium.com>
* res/ael/pval.c, /: Merged revisions 262798 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r262798 |
lmadsen | 2010-05-12 14:53:10 -0500 (Wed, 12 May 2010) | 7 lines
Revert previous WARNING message removal. Marquis42 suggested a
better method of doing what I wanted because I ended up removing
the WARNING message for all instances when really I just wanted
to remove it for the 'return' keyword, not everything. (issue
#17145) ........
* res/ael/pval.c, /: Merged revisions 262796 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r262796 |
lmadsen | 2010-05-12 14:31:42 -0500 (Wed, 12 May 2010) | 4 lines
Remove unnecessary WARNING message in ael/pval.c (closes issue
#17145) Reported by: okrief ........
2010-05-12 18:03 +0000 [r262746] David Vossel <dvossel@digium.com>
* /, apps/app_meetme.c: Merged revisions 262744 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r262744 | dvossel | 2010-05-12 13:01:20 -0500 (Wed, 12 May 2010)
| 17 lines Merged revisions 262662 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r262662 | dvossel | 2010-05-12 12:00:04 -0500 (Wed, 12 May 2010)
| 11 lines fixes app_meetme dsp error We attempted to detect
silence after translating a frame from signed linear. This caused
a flooding of errors. To resolve this the code to detect silence
was moved before the translation. (closes issue #17133) Reported
by: jsdyer ........ ................
2010-05-12 16:29 +0000 [r262516-262659] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_privacy.c: Merged revisions 262656 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r262656 |
tilghman | 2010-05-12 11:23:26 -0500 (Wed, 12 May 2010) | 8 lines
Ensure the arguments are initialized. Also miscellaneous CG
cleanup. (closes issue #16576) Reported by: uxbod Patches:
20100505__issue16576.diff.txt uploaded by tilghman (license 14)
Tested by: uxbod ........
* /, include/asterisk/causes.h: Merged revisions 262513 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r262513 | tilghman | 2010-05-11 16:25:05 -0500 (Tue, 11
May 2010) | 7 lines Move cause 200 to cause 26, as specified in
Q.850. Also cleanup the formatting and add a few more that seem
like good candidates. (closes issue #16157) Reported by: wimpy
........
2010-05-11 19:58 +0000 [r262425] Jason Parker <jparker@digium.com>
* /, res/Makefile: Merged revisions 262422 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r262422 | qwell | 2010-05-11 14:57:24 -0500 (Tue, 11 May 2010) |
18 lines Merged revisions 262421 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r262421 | qwell | 2010-05-11 14:55:42 -0500 (Tue, 11 May 2010) |
11 lines Use a less silly method for modifying a flex-generated
file. The sed syntax that was used wasn't actually valid, causing
some versions to choke. This is the method that is used in 1.6.x+
for similar changes. (closes issue #16696) Reported by: bklang
Patches: 16696-sedfix.diff uploaded by qwell (license 4) Tested
by: qwell ........ ................
2010-05-11 19:41 +0000 [r262415-262420] Paul Belanger <paul.belanger@polybeacon.com>
* pbx/pbx_config.c, /: Merged revisions 262419 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r262419 |
pabelanger | 2010-05-11 15:40:37 -0400 (Tue, 11 May 2010) | 8
lines Improve logging by displaying line number (closes issue
#16303) Reported by: dant Patches: issue16303.patch.v2 uploaded
by pabelanger (license 224) Tested by: dant, lmadsen, pabelanger
........
* /, channels/chan_sip.c: Merged revisions 262414 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r262414 |
pabelanger | 2010-05-11 15:26:17 -0400 (Tue, 11 May 2010) | 8
lines Improve logging information for misconfigured contexts
(closes issue #17238) Reported by: pprindeville Patches:
chan_sip-bug17238.patch uploaded by pprindeville (license 347)
Tested by: pprindeville ........
2010-05-11 17:25 +0000 [r262340] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c, /, Makefile.rules: Merged revisions 262330
via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r262330 | tilghman | 2010-05-11 12:23:51 -0500
(Tue, 11 May 2010) | 9 lines Merged revisions 262321 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r262321 | tilghman | 2010-05-11 12:22:07 -0500 (Tue, 11
May 2010) | 2 lines Fix issue #17302 a slightly different way
(mad props to Qwell) ........ ................
2010-05-10 19:06 +0000 [r262237-262241] David Vossel <dvossel@digium.com>
* /, apps/app_directed_pickup.c: Merged revisions 262240 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r262240 | dvossel | 2010-05-10 14:06:08 -0500 (Mon, 10
May 2010) | 9 lines fixes PickupChan application (closes issue
#16863) Reported by: schern Patches: app_directed_pickup.c.patch
uploaded by schern (license 995) for_trunk.diff uploaded by
cjacobsen (license 1029) Tested by: Graber, cjacobsen, lathama,
rickead2000, dvossel ........
* channels/chan_console.c, /: Merged revisions 262236 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r262236 | dvossel | 2010-05-10 13:36:10 -0500 (Mon, 10 May 2010)
| 11 lines fixes crash in chan_console There is a race condition
between console_hangup() and start_stream(). It is possible for
console_hangup() to be called and then the stream thread to begin
after the hangup. To avoid this a check in start_stream() to make
sure the pvt-owner still exists while the pvt lock is held is
made. If the owner is gone that means the channel hung up and
start_stream should be aborted. ........
2010-05-10 16:39 +0000 [r262155] Tilghman Lesher <tlesher@digium.com>
* /, Makefile.rules: Merged revisions 262152 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r262152 | tilghman | 2010-05-10 11:36:25 -0500 (Mon, 10 May 2010)
| 17 lines Merged revisions 262151 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r262151 | tilghman | 2010-05-10 11:34:21 -0500 (Mon, 10 May 2010)
| 10 lines Allow compilation on Mac OS X 10.4 (Tiger) (closes
issue #17297) Reported by: jcovert Patches:
20100506__issue17297.diff.txt uploaded by tilghman (license 14)
(closes issue #17302) Reported by: jcovert ........
................
2010-05-09 02:17 +0000 [r261916-262105] Tilghman Lesher <tlesher@digium.com>
* autoconf/ast_ext_lib.m4, autoconf/ast_c_compile_check.m4,
autoconf/ast_c_define_check.m4, /, configure,
include/asterisk/autoconfig.h.in: Merged revisions 262102 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r262102 | tilghman | 2010-05-08 21:14:04 -0500 (Sat, 08
May 2010) | 5 lines Cleanup a bit more by getting rid of useless
version defines. Also make library detection use passed CFLAGS.
(closes issue #17309) Reported by: stuarth ........
* /, configure, configure.ac: Merged revisions 262048 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r262048 | tilghman | 2010-05-07 21:40:01 -0500 (Fri, 07 May 2010)
| 2 lines Use CPPFLAGS to pass PTHREAD_CFLAGS for vpb only
........
* /, funcs/func_odbc.c: Merged revisions 261917 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r261917 |
tilghman | 2010-05-07 15:54:35 -0500 (Fri, 07 May 2010) | 8 lines
Double free crash (closes issue #17245) Reported by:
thedavidfactor Patches: 20100426__issue17245.diff.txt uploaded by
tilghman (license 14) Tested by: murraytm ........
* /, configure, include/asterisk/autoconfig.h.in, configure.ac:
Merged revisions 261913 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r261913 |
tilghman | 2010-05-07 15:35:17 -0500 (Fri, 07 May 2010) | 14
lines Use the detected pthread building flags in every place,
instead of hardcoding -lpthread. We nicely detect the right flags
on each system for building Asterisk with pthreads, then ignore
it for every other build option that requires us to build with
pthreads. This caused some items to return a false negative. Also
cleanup some minor naming issues that caused "library library"
redundancy in the output. (closes issue #17303) Reported by:
stuarth Patches: 20100507__issue17303.diff.txt uploaded by
tilghman (license 14) Tested by: stuarth ........
2010-05-07 16:08 +0000 [r261868] Leif Madsen <lmadsen@digium.com>
* UPGRADE-1.6.txt, /: Merged revisions 261867 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r261867 |
lmadsen | 2010-05-07 11:05:24 -0500 (Fri, 07 May 2010) | 6 lines
Update UPGRADE-1.6.txt stating insecure=very has been removed.
(closes issue #17282) Reported by: stuarth Tested by: stuarth
........
2010-05-06 20:13 +0000 [r261739] Jeff Peeler <jpeeler@digium.com>
* apps/app_voicemail.c, /: Merged revisions 261736 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r261736 | jpeeler | 2010-05-06 15:11:53 -0500
(Thu, 06 May 2010) | 15 lines Merged revisions 261735 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r261735 | jpeeler | 2010-05-06 15:10:59 -0500 (Thu, 06 May 2010)
| 8 lines Only allow the operator key to be accepted after
leaving a voicemail. Or rather disallow the operator key from
being accepted when not offered, such as after finishing a
recording from within the mailbox options menu. ABE-2121 SWP-1267
........ ................
2010-05-06 17:08 +0000 [r261612] Jason Parker <jparker@digium.com>
* sounds/Makefile, /: Merged revisions 261609 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r261609 | qwell | 2010-05-06 12:06:40 -0500 (Thu, 06 May 2010) |
11 lines Merged revisions 261608 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r261608 | qwell | 2010-05-06 11:56:02 -0500 (Thu, 06 May 2010) |
4 lines Use the versioned MOH tarballs, now that we have them.
This makes for more reproducibility. Prompted by a discussion in
#asterisk-dev ........ ................
2010-05-06 15:43 +0000 [r261563] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_sip.c: Merged revisions 261560 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r261560 |
tilghman | 2010-05-06 10:39:10 -0500 (Thu, 06 May 2010) | 8 lines
Permit more lines within a SIP body to be parsed. The example
given within the related issue showed 120 lines, which was mostly
a result of the body being XML. (closes issue #17179) Reported
by: khw ........
2010-06-01 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.8 Released.
2010-05-26 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.8-rc2 Released.
2010-05-26 10:56 -0500 [r265891] Matt Nicholson <mnicholson@digium.com>
* Merged r265610 from 1.4:
Don't mark the cdr records of unanswered queue calls with "NOANSWER".
This restores the behavior prior to r258670.
(closes issue #17334)
Reported by: jvandal
Patches:
queue-cdr-fixes1.diff uploaded by mnicholson (license 96)
Tested by: aragon, jvandal
2010-05-06 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.8-rc1 Released
2010-05-06 14:07 +0000 [r261498-261499] Russell Bryant <russell@digium.com>
* tests/test_heap.c: Add test case that ensures the heap handles
arbitrary removals properly. (issue #17277) Reported by:
cappucinoking Patches: test_heap.diff uploaded by cappucinoking
(license 1036) Tested by: cappucinoking, russell
* /, main/heap.c: Merged revisions 261496 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r261496 |
russell | 2010-05-06 08:58:07 -0500 (Thu, 06 May 2010) | 40 lines
Fix handling of removing nodes from the middle of a heap. This
bug surfaced in 1.6.2 and does not affect code in any other
released version of Asterisk. It manifested itself as SIP qualify
not happening when it should, causing peers to go unreachable.
This was debugged down to scheduler entries sometimes not getting
executed when they were supposed to, which was in turn caused by
an error in the heap code. The problem only sometimes occurs, and
it is due to the logic for removing an entry in the heap from an
arbitrary location (not just popping off the top). The scheduler
performs this operation frequently when entries are removed
before they run (when ast_sched_del() is used). In a normal pop
off of the top of the heap, a node is taken off the bottom,
placed at the top, and then bubbled down until the max heap
property is restored (see max_heapify()). This same logic was
used for removing an arbitrary node from the middle of the heap.
Unfortunately, that logic is full of fail. This patch fixes that
by fully restoring the max heap property when a node is thrown
into the middle of the heap. Instead of just pushing it down as
appropriate, it first pushes it up as high as it will go, and
_then_ pushes it down. Lastly, fix a minor problem in
ast_heap_verify(), which is only used for debugging. If a parent
and child node have the same value, that is not an error. The
only error is if a parent's value is less than its children. A
huge thanks goes out to cappucinoking for debugging this down to
the scheduler, and then producing an ast_heap test case that
demonstrated the breakage. That made it very easy for me to focus
on the heap logic and produce a fix. Open source projects are
awesome. (closes issue #16936) Reported by: ib2 Tested by:
cappucinoking, crjw (closes issue #17277) Reported by:
cappucinoking Patches: heap-fix.rev2.diff uploaded by russell
(license 2) Tested by: cappucinoking, russell ........
2010-05-06 07:43 +0000 [r261453] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* channels/chan_dahdi.c, /: Merged revisions 261451 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r261451 | tzafrir | 2010-05-06 10:27:31 +0300 (ה', 06 מאי 2010) |
4 lines When failing to configure, don't destroy 'cfg' twice
Fixes a crash when some config section had an incorrect channel
config. ........
2010-05-05 19:08 +0000 [r261233-261315] Paul Belanger <paul.belanger@polybeacon.com>
* /, channels/chan_sip.c: Merged revisions 261314 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r261314 | pabelanger | 2010-05-05 14:43:03 -0400 (Wed, 05 May
2010) | 19 lines Merged revisions 261274 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r261274 | pabelanger | 2010-05-05 12:42:22 -0400 (Wed, 05 May
2010) | 12 lines Registration fix for SIP realtime. Make sure
realtime fields are not empty. (closes issue #17266) Reported by:
Nick_Lewis Patches: chan_sip.c-realtime.patch uploaded by Nick
Lewis (license 657) Tested by: Nick_Lewis, sberney Review:
https://reviewboard.asterisk.org/r/643/ ........ ................
* apps/app_queue.c, /: Merged revisions 261232 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r261232 |
pabelanger | 2010-05-05 11:42:07 -0400 (Wed, 05 May 2010) | 10
lines 'queue reset stats' erroneously clears wrapuptime
configuration. Resets each member's lastcall to 0 now. (closes
issue #17262, #16519) Reported by: rain Patches:
wrapuptime_reset_fix.diff uploaded by rain (license 327) Tested
by: rain ........
2010-05-04 23:55 +0000 [r261098] Tilghman Lesher <tlesher@digium.com>
* main/channel.c, /: Merged revisions 261095 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r261095 | tilghman | 2010-05-04 18:51:52 -0500 (Tue, 04 May 2010)
| 18 lines Merged revisions 261093-261094 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r261093 | tilghman | 2010-05-04 18:36:53 -0500 (Tue, 04 May 2010)
| 7 lines Protect against overflow, when calculating how long to
wait for a frame. (closes issue #17128) Reported by: under
Patches: d.diff uploaded by under (license 914) ........ r261094
| tilghman | 2010-05-04 18:47:08 -0500 (Tue, 04 May 2010) | 2
lines Add a tiny corner case to the previous commit ........
................
2010-05-04 19:01 +0000 [r260927] Jeff Peeler <jpeeler@digium.com>
* apps/app_voicemail.c, /: Merged revisions 260924 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r260924 | jpeeler | 2010-05-04 13:51:28 -0500
(Tue, 04 May 2010) | 18 lines Merged revisions 260923 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r260923 | jpeeler | 2010-05-04 13:46:46 -0500 (Tue, 04 May 2010)
| 12 lines Voicemail transfer to operator should occur
immediately, not after main menu. There were two scenarios in the
advanced options that while using the operator=yes and review=yes
options, the transfer occurred only after exiting the main menu
(after sending a reply or leaving a message for an extension).
Now after the audio is processed for the reply or message the
transfer occurs immediately as expected. ABE-2107 ABE-2108
........ ................
2010-05-04 16:58 +0000 [r260884] Matthew Nicholson <mnicholson@digium.com>
* configs/sip.conf.sample, include/asterisk/frame.h,
main/channel.c, /, channels/chan_sip.c: Merged revisions 254450
via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r254450 | kpfleming | 2010-03-25 10:27:31 -0500 (Thu, 25
Mar 2010) | 49 lines Improve handling of T.38 re-INVITEs that
arrive before a T.38-capable application is executing on a
channel. This patch addresses an issue found during working with
end-users using res_fax. If an incoming call is answered in the
dialplan, or jumps to the 'fax' extension due to reception of a
CNG tone (with faxdetect enabled), and then the remote endpoint
sends a T.38 re-INVITE, it is possible for the channel's T.38
state to be 'T38_STATE_NEGOTIATING' when the application starts
up. Unfortunately, even if the application wants to use T.38, it
can't respond to the peer's negotiation request, because the
AST_CONTROL_T38_PARAMETERS control frame that chan_sip sent
originally has been lost, and the application needs the content
of that frame to be able to formulate a reply. This patch adds a
new 'request' type to AST_CONTROL_T38_PARAMETERS,
AST_T38_REQUEST_PARMS. If the application sends this request,
chan_sip will re-send the original control frame (with
AST_T38_REQUEST_NEGOTIATE as the request type), and the
application can respond as normal. If this occurs within the five
second timeout in chan_sip, the automatic cancellation of the
peer reinvite will be stopped, and the application will 'own' the
negotiation process from that point onwards. This also improves
the code path in chan_sip to allow sip_indicate(), when called
for AST_CONTROL_T38_PARAMETERS, to be able to return a non-zero
response, which should have been in place before since the
control frame *can* fail to be processed properly. It also
modifies ast_indicate() to return whatever result the channel
driver returned for this control frame, rather than converting
all non-zero results into '-1'. Finally, the new request type
intentionally returns a positive value, so that an application
that sends AST_T38_REQUEST_PARMS can know for certain whether the
channel driver accepted it and will be replying with a control
frame of its own, or whether it was ignored (if the
sip_indicate()/ast_indicate() path had properly supported failure
responses before, this would not be necessary). This patch also
modifies res_fax to take advantage of the new request. In
addition, this patch makes sip_t38_abort() actually lock the
private structure before doing its work... bad programmer, no
donut. This patch also enhances chan_sip's 'faxdetect' support to
allow triggering on T.38 re-INVITEs received as well as CNG tone
detection. Review: https://reviewboard.asterisk.org/r/556/
........
2010-05-04 15:51 +0000 [r260746-260805] Jason Parker <jparker@digium.com>
* /, build_tools/make_build_h: Merged revisions 260802 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r260802 | qwell | 2010-05-04 10:49:57 -0500
(Tue, 04 May 2010) | 9 lines Merged revisions 260801 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r260801 | qwell | 2010-05-04 10:49:27 -0500 (Tue, 04 May
2010) | 1 line Fix fallout from removing from configure script.
Pointed out by philipp64 on #asterisk-dev ........
................
* /: Fix merge props
2010-05-03 17:42 +0000 [r260743] Paul Belanger <paul.belanger@polybeacon.com>
* Makefile, /: Merged revisions 260661-260662 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r260661 | pabelanger | 2010-05-03 12:41:30 -0400 (Mon, 03 May
2010) | 10 lines non-root make install PREFIX=/tmp fails. Prepend
libdir when executing mkpkgconfig allowing non-root installs to
work. (closes issue #17268) Reported by: pabelanger Patches:
issue17268.patch uploaded by pabelanger (license 224) Tested by:
pabelanger ........ r260662 | pabelanger | 2010-05-03 12:54:41
-0400 (Mon, 03 May 2010) | 3 lines Should have removed /usr/lib/
part. Thanks Qwell. ........
2010-05-03 14:59 +0000 [r260571] Leif Madsen <lmadsen@digium.com>
* doc/HOWTO_collect_debug_information.txt: Merged revisions 260570
via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r260570 | lmadsen | 2010-05-03 09:58:23 -0500
(Mon, 03 May 2010) | 9 lines Merged revisions 260569 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r260569 | lmadsen | 2010-05-03 09:57:39 -0500 (Mon, 03
May 2010) | 1 line Minor typo pointed out by pabelanger on IRC.
........ ................
2010-04-30 22:48 +0000 [r260441] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 260437 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r260437 | jpeeler | 2010-04-30 17:36:49 -0500
(Fri, 30 Apr 2010) | 18 lines Merged revisions 260434 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r260434 | jpeeler | 2010-04-30 17:22:46 -0500 (Fri, 30 Apr 2010)
| 11 lines Ensure channel state is not incorrectly set in the
case of a very early answer. The needringing bit was being read
in dahdi_read after answering thereby setting the state to
ringing from up. This clears needringing upon answering so that
is no longer possible. (closes issue #17067) Reported by: tzafrir
Patches: needringing.diff uploaded by tzafrir (license 46)
........ ................
2010-04-30 20:22 +0000 [r260373] Mark Michelson <mmichelson@digium.com>
* res/res_musiconhold.c, /: Merged revisions 260346 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r260346 | mmichelson | 2010-04-30 15:11:02 -0500
(Fri, 30 Apr 2010) | 24 lines Merged revisions 260345 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r260345 | mmichelson | 2010-04-30 15:08:15 -0500 (Fri, 30 Apr
2010) | 18 lines Fix potential crash from race condition due to
accessing channel data without the channel locked. In
res_musiconhold.c, there are several places where a channel's
stream's existence is checked prior to calling ast_closestream on
it. The issue here is that in several cases, the channel was not
locked while checking the stream. The result was that if two
threads checked the state of the channel's stream at
approximately the same time, then there could be a situation
where both threads attempt to call ast_closestream on the
channel's stream. The result here is that the refcount for the
stream would go below 0, resulting in a crash. I have added
proper channel locking to res_musiconhold.c to ensure that we do
not try to check chan->stream without the channel locked. A
Digium customer has been using this patch for several weeks and
has not had any crashes since applying the patch. ABE-2147
........ ................
2010-04-30 06:22 +0000 [r260281-260303] Tilghman Lesher <tlesher@digium.com>
* /, main/app.c: Merged revisions 260292 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r260292 |
tilghman | 2010-04-30 01:19:35 -0500 (Fri, 30 Apr 2010) | 13
lines Don't allow file descriptors to go above 64k, when we're
closing them in a fork(2). This saves time, when, even though the
system allows the process limit to be that high, the practical
limit is much lower. (closes issue #17223) Reported by:
dbackeberg Patches: 20100423__issue17223.diff.txt uploaded by
tilghman (license 14) Tested by: dbackeberg ........
* configs/extensions.conf.sample, /: Merged revisions 260280 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r260280 | tilghman | 2010-04-30 00:23:56 -0500 (Fri, 30
Apr 2010) | 7 lines Logic fixups for a sample FREENUM dialplan
context. (closes issue #17263) Reported by: pprindeville Patches:
freenum-dialplan.patch#3 uploaded by pprindeville (license 347)
........
2010-04-29 23:13 +0000 [r260234] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 260231 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r260231 | rmudgett | 2010-04-29 17:44:14 -0500
(Thu, 29 Apr 2010) | 33 lines Merged revisions 260195 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r260195 | rmudgett | 2010-04-29 17:11:47 -0500 (Thu, 29 Apr 2010)
| 26 lines DTMF CallerID detection problems. The code handling
DTMF CallerID drops digits on long CallerID numbers and may
timeout waiting for the first ring with shorter numbers. The DTMF
emulation mode was not turned off when processing DTMF CallerID.
When the emulation code gets behind in processing the DTMF digits
it can skip a digit. For shorter numbers, the timeout may have
been too short. I increased it from 2 seconds to 4 seconds. Four
seconds is a typical time between rings for many countries.
(closes issue #16460) Reported by: sum Patches: issue16460.patch
uploaded by rmudgett (license 664) issue16460_v1.6.2.patch
uploaded by rmudgett (license 664) Tested by: sum, rmudgett
Review: https://reviewboard.asterisk.org/r/634/ JIRA SWP-562 JIRA
AST-334 JIRA SWP-901 ........ ................
2010-04-29 18:18 +0000 [r260156] Tilghman Lesher <tlesher@digium.com>
* configs/extensions.conf.sample, /: Merged revisions 260148 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r260148 | tilghman | 2010-04-29 13:15:57 -0500 (Thu, 29
Apr 2010) | 2 lines Pattern match fail. ........
2010-04-29 15:35 +0000 [r260051] David Vossel <dvossel@digium.com>
* main/audiohook.c, /, include/asterisk/audiohook.h: Merged
revisions 260050 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r260050 | dvossel | 2010-04-29 10:33:27 -0500 (Thu, 29 Apr 2010)
| 21 lines Merged revisions 260049 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r260049 | dvossel | 2010-04-29 10:31:02 -0500 (Thu, 29 Apr 2010)
| 14 lines Fixes crash in audiohook_write_list The middle_frame
in the audiohook_write_list function was being freed if a
audiohook manipulator returned a failure. This is incorrect
logic. This patch resolves this and adds detailed descriptions of
how this function should work and why manipulator failures must
be ignored. (closes issue #17052) Reported by: dvossel Tested by:
dvossel (closes issue #16196) Reported by: atis Review:
https://reviewboard.asterisk.org/r/623/ ........ ................
2010-04-28 22:36 +0000 [r259959] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 259957 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r259957 |
mmichelson | 2010-04-28 17:34:15 -0500 (Wed, 28 Apr 2010) | 11
lines Don't override peer context with domain context. (closes
issue #17040) Reported by: pprindeville Patches:
asterisk-1.6-bugid17040.patch uploaded by pprindeville (license
347) Tested by: pprindeville Review:
https://reviewboard.asterisk.org/r/565/ ........
2010-04-28 21:26 +0000 [r259899] David Vossel <dvossel@digium.com>
* main/channel.c, channels/chan_local.c, /: Merged revisions 259870
via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r259870 | dvossel | 2010-04-28 16:20:03 -0500
(Wed, 28 Apr 2010) | 39 lines Merged revisions 259858 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r259858 | dvossel | 2010-04-28 16:16:03 -0500 (Wed, 28 Apr 2010)
| 33 lines resolves deadlocks in chan_local Issue_1. In the
local_hangup() 3 locks must be held at the same time... pvt,
pvt->chan, and pvt->owner. Proper deadlock avoidance is done when
the channel to hangup is the outbound chan_local channel, but
when it is not the outbound channel we have an issue... We
attempt to do deadlock avoidance only on the tech pvt, when both
the tech pvt and the pvt->owner are locked coming into that loop.
By never giving up the pvt->owner channel deadlock avoidance is
not entirely possible. This patch resolves that by doing deadlock
avoidance on both the pvt->owner and the pvt when trying to get
the pvt->chan lock. Issue_2. ast_prod() is used in
ast_activate_generator() to queue a frame on the channel and make
the channel's read function get called. This function is used in
ast_activate_generator() while the channel is locked, which
mean's the channel will have a lock both from the generator code
and the frame_queue code by the time it gets to chan_local.c's
local_queue_frame code... local_queue_frame contains some of the
same crazy deadlock avoidance that local_hangup requires, and
this recursive lock prevents that deadlock avoidance from
happening correctly. This patch removes ast_prod() from the
channel lock so only one lock is held during the
local_queue_frame function. (closes issue #17185) Reported by:
schmoozecom Patches: issue_17185_v1.diff uploaded by dvossel
(license 671) issue_17185_v2.diff uploaded by dvossel (license
671) Tested by: schmoozecom, GameGamer43 Review:
https://reviewboard.asterisk.org/r/631/ ........ ................
2010-04-28 21:09 +0000 [r259854] Leif Madsen <lmadsen@digium.com>
* config.guess: Merged revisions 259853 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r259853 | lmadsen | 2010-04-28 16:08:34 -0500 (Wed, 28 Apr 2010)
| 14 lines Merged revisions 259852 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r259852 | lmadsen | 2010-04-28 16:07:48 -0500 (Wed, 28 Apr 2010)
| 6 lines Update config.guess. Updating config.guess because
after installing Ubuntu Server 9.10 and running all the update
scripts, running ./configure would not continue because it was
unable to determine what kind of system I had. After updating
config.guess things started working again. ........
................
2010-04-28 20:34 +0000 [r259781-259851] Jason Parker <jparker@digium.com>
* /, configure, configure.ac: Merged revisions 259848 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r259848 | qwell | 2010-04-28 15:32:14 -0500
(Wed, 28 Apr 2010) | 9 lines Merged revisions 259847 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r259847 | qwell | 2010-04-28 15:30:21 -0500 (Wed, 28 Apr
2010) | 1 line Add AC_CONFIG_AUX_DIR to configure script, so
systems without install can use install-sh from our source dir.
........ ................
* makeopts.in, /: Merged revisions 259837 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r259837 | qwell | 2010-04-28 15:26:35 -0500 (Wed, 28 Apr 2010) |
9 lines Merged revisions 259833 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r259833 | qwell | 2010-04-28 15:25:36 -0500 (Wed, 28 Apr 2010) |
1 line Missed this when removing $ID ........ ................
* Makefile, /, configure, configure.ac: Merged revisions 259760 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r259760 | qwell | 2010-04-28 14:19:54 -0500
(Wed, 28 Apr 2010) | 14 lines Merged revisions 259748 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r259748 | qwell | 2010-04-28 14:17:38 -0500 (Wed, 28 Apr 2010) |
7 lines Remove usage of `id` since it isn't useful and was
causing breakge. Solaris `id` doesn't support the -u argument.
Instead of figuring out how to fix this to work on Solaris, I
decided to check why it was necessary and where else it was used.
It was only used in one place, and it hasn't been needed for a
very long time (I question whether it was ever needed). ........
................
2010-04-28 17:19 +0000 [r259681] Jeff Peeler <jpeeler@digium.com>
* apps/app_voicemail.c, /: Merged revisions 259672 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r259672 | jpeeler | 2010-04-28 12:18:43 -0500
(Wed, 28 Apr 2010) | 11 lines Merged revisions 259664 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r259664 | jpeeler | 2010-04-28 12:13:29 -0500 (Wed, 28 Apr 2010)
| 4 lines Do not play goodbye prompt after timeout of message
review. ABE-2124 ........ ................
2010-04-27 22:46 +0000 [r259616] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 259538 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r259538 | rmudgett | 2010-04-27 17:18:09 -0500
(Tue, 27 Apr 2010) | 18 lines Merged revisions 259531 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r259531 | rmudgett | 2010-04-27 16:53:07 -0500 (Tue, 27 Apr 2010)
| 11 lines DAHDI "WARNING" message is confusing and vague
"WARNING[28406]: chan_dahdi.c:6873 ss_thread: CallerID feed
failed: Success" Changed the warning to "Failed to decode
CallerID on channel 'name'". The message before it is likely more
specific about why the CallerID decode failed. SWP-501 AST-283
........ ................
2010-04-27 21:50 +0000 [r259528] Leif Madsen <lmadsen@digium.com>
* sounds/Makefile: Merged revisions 259527 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r259527 | lmadsen | 2010-04-27 16:49:36 -0500 (Tue, 27 Apr 2010)
| 23 lines Merged revisions 259526 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r259526 | lmadsen | 2010-04-27 16:48:47 -0500 (Tue, 27 Apr 2010)
| 15 lines Update sounds files. * Add additional sounds prompts
for say_enumeration * Update the English conference sounds
prompts so they are better quality and all sound more consistent
* Clean up the core-sounds-XX.txt and extra-sounds-XX.txt files
to include all present sound files Both core (en, fr, es) and
extra (en, fr) sounds files have been updated. (closes issue
#16200) Reported by: murf (closes issue #17137) Reported by:
lmadsen ........ ................
2010-04-27 21:25 +0000 [r259356-259486] Jason Parker <jparker@digium.com>
* main/editline/configure.in, /, main/editline/configure,
main/editline/Makefile.in: Merged revisions 259439 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r259439 | qwell | 2010-04-27 16:13:01 -0500 (Tue, 27 Apr 2010) |
5 lines Add gar to the check for AR for those silly OSes
(Solaris) that don't have ar. autoconf2.13 couldn't handle
AC_PROG_GREP, so I removed it. This is fine, since we don't need
to use anything that the configure script doesn't. ........
* /: Unblock revision 259439.
* /, configure, configure.ac: Merged revisions 259353 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r259353 | qwell | 2010-04-27 14:31:55 -0500
(Tue, 27 Apr 2010) | 12 lines Merged revisions 259352 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r259352 | qwell | 2010-04-27 14:29:26 -0500 (Tue, 27 Apr 2010) |
5 lines Support the silly OSes that don't have ar and strip.
Since AC_PATH_TOOL is equiv to AC_CHECK_TOOL when path isn't
specified, and AC_PATH_TOOLS doesn't exist, we'll just switch to
AC_CHECK_TOOLS. ........ ................
2010-04-27 19:03 +0000 [r259310] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged
revisions 259307 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r259307 | rmudgett | 2010-04-27 13:29:33 -0500 (Tue, 27 Apr 2010)
| 21 lines Merged revisions 259270 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r259270 | rmudgett | 2010-04-27 13:14:54 -0500 (Tue, 27 Apr 2010)
| 14 lines hidecalleridname parameter in chan_dahdi.conf Issue
#7321 implements a new chan_dahdi configuration option. However,
a change mentioned in the issue was never implemented. This is
the change that will allow the feature to work. I added a note to
chan_dahdi.conf.sample about the feature. (closes issue #17143)
Reported by: djensen99 Patches: diff.txt uploaded by djensen99
(license NA) (One line change) Tested by: djensen99 ........
................
2010-04-26 21:48 +0000 [r259103-259109] Mark Michelson <mmichelson@digium.com>
* main/channel.c, /: Merged revisions 259105 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r259105 | mmichelson | 2010-04-26 16:45:13 -0500 (Mon, 26 Apr
2010) | 9 lines Merged revisions 259104 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r259104 | mmichelson | 2010-04-26 16:44:43 -0500 (Mon, 26 Apr
2010) | 3 lines Let compilation succeed warning-free when
DONT_OPTIMIZE is turned off. ........ ................
* main/channel.c, /: Merged revisions 259023 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r259023 | mmichelson | 2010-04-26 16:13:35 -0500 (Mon, 26 Apr
2010) | 19 lines Merged revisions 259018 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r259018 | mmichelson | 2010-04-26 16:03:08 -0500 (Mon, 26 Apr
2010) | 13 lines Prevent Newchannel manager events for dummy
channels. No Newchannel manager event will be fired for channels
that are allocated to not match a registered technology type.
Thus bogus channels allocated solely for variable substitution or
CDR operations do not result in a Newchannel event. (closes issue
#16957) Reported by: atis Review:
https://reviewboard.asterisk.org/r/601 ........ ................
2010-04-26 16:00 +0000 [r258935] Leif Madsen <lmadsen@digium.com>
* /, channels/chan_sip.c: Merged revisions 258934 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r258934 |
lmadsen | 2010-04-26 10:59:34 -0500 (Mon, 26 Apr 2010) | 7 lines
Small error in the T.140 RTP port verbose log. (closes issue
#16988) Reported by: frawd Patches: chan_sip_sdp_verbose_fix.diff
uploaded by frawd (license 610) Tested by: russell ........
2010-04-25 18:14 +0000 [r258779] Tilghman Lesher <tlesher@digium.com>
* res/res_monitor.c, /: Merged revisions 258776 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r258776 | tilghman | 2010-04-25 13:12:14 -0500 (Sun, 25 Apr 2010)
| 13 lines Merged revisions 258775 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r258775 | tilghman | 2010-04-25 13:09:05 -0500 (Sun, 25 Apr 2010)
| 6 lines When StopMonitor is called, ensure that it will not be
restarted by a channel event. (closes issue #16590) Reported by:
kkm Patches: resmonitor-16590-trunk.239289.diff uploaded by kkm
(license 888) ........ ................
2010-04-22 22:15 +0000 [r258676] Matthew Nicholson <mnicholson@digium.com>
* main/cdr.c, main/channel.c, /, main/features.c: Merged revisions
258671,258675 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r258671 | mnicholson | 2010-04-22 16:57:59 -0500 (Thu, 22 Apr
2010) | 32 lines Merged revisions 193391,258670 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r193391 | mnicholson | 2009-05-08 16:01:25 -0500 (Fri, 08 May
2009) | 8 lines Set the proper disposition on originated calls.
(closes issue #14167) Reported by: jpt Patches:
call-file-missing-cdr2.diff uploaded by mnicholson (license 96)
Tested by: dlotina, rmartinez, mnicholson ........ r258670 |
mnicholson | 2010-04-22 16:49:07 -0500 (Thu, 22 Apr 2010) | 11
lines Fix broken CDR behavior. This change allows a CDR record
previously marked with disposition ANSWERED to be set as BUSY or
NO ANSWER. Additionally this change partially reverts r235635 and
does not set the AST_CDR_FLAG_ORIGINATED flag on CDRs generated
from ast_call(). To preserve proper CDR behavior, the
AST_CDR_FLAG_DIALED flag is now cleared from all brige CDRs in
ast_bridge_call(). (closes issue #16797) Reported by:
VarnishedOtter Tested by: mnicholson ........ (closes issue
#16222) Reported by: telles Tested by: mnicholson
................ r258675 | mnicholson | 2010-04-22 17:11:23 -0500
(Thu, 22 Apr 2010) | 2 lines Fix previous commit.
................
2010-04-22 21:58 +0000 [r258516-258672] Russell Bryant <russell@digium.com>
* /, main/event.c: Merged revisions 258632 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk For 1.6.2, only
merge the bug fixes, not the unit test. ........ r258632 |
russell | 2010-04-22 16:06:53 -0500 (Thu, 22 Apr 2010) | 22 lines
Add ast_event subscription unit test and fix some ast_event API
bugs. This patch introduces another test in test_event.c that
exercises most of the subscription related ast_event API calls. I
made some minor additions to the existing event allocation test
to increase API coverage by the test code. Finally, I made a list
in a comment of API calls not yet touched by the test module as a
to-do list for future test development. During the development of
this test code, I discovered a number of bugs in the event API.
1) subscriptions to AST_EVENT_ALL were not handled appropriately
in a couple of different places. The API allows a subscription to
all event types, but with IE parameters, just as if it was a
subscription to a specific event type. However, the parameters
were being ignored. This affected ast_event_check_subscriber()
and event distribution to subscribers. 2) Some of the logic in
ast_event_check_subscriber() for checking subscriptions against
query parameters was wrong. Review:
https://reviewboard.asterisk.org/r/617/ ........
* /, doc/tex/channelvariables.tex: Merged revisions 258515 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r258515 | russell | 2010-04-22 12:36:34 -0500 (Thu, 22
Apr 2010) | 2 lines Add MEETMEBOOKID from r256019. ........
2010-04-21 22:11 +0000 [r258436] Jeff Peeler <jpeeler@digium.com>
* apps/app_voicemail.c, /: Merged revisions 258433 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r258433 | jpeeler | 2010-04-21 16:56:09 -0500
(Wed, 21 Apr 2010) | 15 lines Merged revisions 258432 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r258432 | jpeeler | 2010-04-21 16:45:36 -0500 (Wed, 21 Apr 2010)
| 8 lines Fix looping forever when no input received in certain
voicemail menu scenarios. Specifically, prompting for an
extension (when leaving or forwarding a message) or when
prompting for a digit (when saving a message or changing
folders). ABE-2122 SWP-1268 ........ ................
2010-04-21 19:44 +0000 [r258384-258386] Leif Madsen <lmadsen@digium.com>
* doc/tex/asterisk.tex: Remove missed line in previous merge.
(issue #17220)
* configure: Forgot to merge the updated configure script. (issue
#17220)
* doc/tex/localchannel.tex, doc/tex/enum.tex, makeopts.in,
doc/tex/asterisk.tex, Makefile, /, doc/tex/Makefile,
configure.ac, doc/tex/phoneprov.tex, doc/tex, doc/tex/ael.tex,
build_tools/prep_tarball: Merged revisions 258351 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r258351 | lmadsen | 2010-04-21 14:18:35 -0500 (Wed, 21 Apr 2010)
| 20 lines Add ability to generate ASCII documentation from the
TeX files. These changes add the ability to run 'make
asterisk.txt' just like the existing 'make asterisk.pdf' commands
to generate a text document from the TeX files we have in the
doc/tex/ directory. I've also updated a few of the .tex files
because they weren't properly escaping certain characters so they
would show up as Unicode characters (like [U+021C]). Made changes
to the configure scripts so it would detect the catdvi program
which is required to convert the .dvi file generated by latex.
I've also added a few lines to the build_tools/prep_tarball
script so that the text documentation gets generated and added to
future tarballs of Asterisk releases. (closes issue #17220)
Reported by: lmadsen Patches: asterisk.txt.patch uploaded by
lmadsen (license 10) asterisk.txt.patch-v4 uploaded by pabelanger
(license 224) Tested by: lmadsen, pabelanger ........
2010-04-21 18:19 +0000 [r258314] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 258305 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r258305 |
dvossel | 2010-04-21 13:13:36 -0500 (Wed, 21 Apr 2010) | 12 lines
fixes issue with double "sip:" in header field This is a clear
mistake in logic. Future discussions about how to avoid having to
handle uri's like this should take place in the future, but this
fix needs to go in for now. (closes issue #15847) Reported by:
ebroad Patches: doublesip.patch uploaded by ebroad (license 878)
........
2010-04-20 19:03 +0000 [r258148-258150] Leif Madsen <lmadsen@digium.com>
* /, configs/cli_aliases.conf.sample: Merged revisions 258149 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r258149 | lmadsen | 2010-04-20 14:02:49 -0500 (Tue, 20
Apr 2010) | 1 line Add 'soft hangup' alias per Steve Johnson on
asterisk-users. ........
* configs/extensions.conf.sample, /: Merged revisions 258147 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r258147 | lmadsen | 2010-04-20 13:38:39 -0500 (Tue, 20
Apr 2010) | 8 lines Add example dialplan for dialing ISN numbers
(http://www.freenum.org). Minor tweaks and documentation added by
me. (closes issue #17058) Reported by: pprindeville Patches:
freenum.patch#5 uploaded by pprindeville (license 347) Tested by:
lmadsen ........
2010-04-20 18:04 +0000 [r258108] Jeff Peeler <jpeeler@digium.com>
* apps/app_voicemail.c, /: Merged revisions 258065 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r258065 | jpeeler | 2010-04-20 12:06:19 -0500
(Tue, 20 Apr 2010) | 17 lines Merged revisions 258029 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r258029 | jpeeler | 2010-04-20 11:16:33 -0500 (Tue, 20 Apr 2010)
| 11 lines Play correct prompt when voicemail store failure
occurs after attempted forward. If a user's mailbox was full and
a message was attempted to be forwarded to said box, warnings on
the console would indicate failure. However, the played prompt
was that of success (vm-msgsaved). Now storage failure is taken
into account and the correct prompt (vm-mailboxfull) is played
when appropriate. ABE-2123 SWP-1262 ........ ................
2010-04-20 18:02 +0000 [r258107] Leif Madsen <lmadsen@digium.com>
* contrib/scripts/sip-friends.sql, /: Merged revisions 258106 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r258106 | lmadsen | 2010-04-20 13:01:28 -0500 (Tue, 20
Apr 2010) | 7 lines Add missing 'useragent' field to
sip-friends.sql file. (closes issue #17171) Reported by: thehar
Patches: sip-friends.patch uploaded by thehar (license 831)
Tested by: pabelanger, thehar ........
2010-04-19 21:58 +0000 [r257948-257950] Jason Parker <jparker@digium.com>
* main/indications.c, /: Merged revisions 257949 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r257949 |
qwell | 2010-04-19 16:57:56 -0500 (Mon, 19 Apr 2010) | 1 line
Change log message to match severity. ........
* main/indications.c, /: Merged revisions 257947 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r257947 |
qwell | 2010-04-19 16:49:30 -0500 (Mon, 19 Apr 2010) | 6 lines
Don't consider a missing indications.conf to be a critical error.
There were many changes in revision 176627 which would avoid the
error that a missing config would have caused. Other than this,
there are no other config files (including asterisk.conf,
surprisingly) that are required. ........
2010-04-19 18:30 +0000 [r257850] Terry Wilson <twilson@digium.com>
* /, main/features.c: Merged revisions 257810 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r257810 |
twilson | 2010-04-19 12:57:41 -0500 (Mon, 19 Apr 2010) | 5 lines
Fix incomplete CDR merge from r195881 Because res/res_features.c
was removed and main/cdr.c added, these changes didn't make it to
trunk and the 1.6.x branches ........
2010-04-18 17:28 +0000 [r257771] Tilghman Lesher <tlesher@digium.com>
* configs/cdr_odbc.conf.sample, /: Merged revisions 257768 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r257768 | tilghman | 2010-04-18 12:25:53 -0500 (Sun, 18
Apr 2010) | 2 lines Removing unused configuration parameters
........
2010-04-16 21:47 +0000 [r257740] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
* apps/app_mixmonitor.c, /: Merged revisions 257713 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r257713 | dhubbard | 2010-04-16 16:22:30 -0500
(Fri, 16 Apr 2010) | 28 lines Merged revisions 257686 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r257686 | dhubbard | 2010-04-16 16:15:43 -0500 (Fri, 16 Apr 2010)
| 21 lines Make the mixmonitor thread process audio frames faster
Mantis issue 17078 reports MixMonitor recordings have shorter
durations than the call duration. This was because the mixmonitor
thread was not processing frames from the audiohook fast enough.
The mixmonitor thread would slowly fall behind the most recent
audio frame and when the channel hangs up, the mixmonitor thread
would exit without processing the same number of frames as the
channel; leaving the mixmonitor recording shorter than actual
call duration. This revision fixes this issue by moving the
ast_audiohook_trigger_wait() and the subsequent audiohook.status
check into the block where the ast_audiohook_read_frame()
function returns NULL. (closes issue #17078) Reported by:
geoff2010 Patches: dw-M17078.patch uploaded by dhubbard (license
733) Tested by: dhubbard, geoff2010 Review:
https://reviewboard.asterisk.org/r/611/ ........ ................
2010-04-15 21:34 +0000 [r257510-257597] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/app.h, /, main/app.c: Merged revisions 257560
via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r257560 | tilghman | 2010-04-15 16:26:19 -0500
(Thu, 15 Apr 2010) | 13 lines Merged revisions 257544 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r257544 | tilghman | 2010-04-15 16:23:24 -0500 (Thu, 15 Apr 2010)
| 6 lines Allow application options with arguments to contain
parentheses, through a variety of escaping techniques. Fixes
SWP-1194 (ABE-2143). Review:
https://reviewboard.asterisk.org/r/604/ ........ ................
* /, channels/chan_sip.c: Merged revisions 257493 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r257493 | tilghman | 2010-04-15 15:30:15 -0500 (Thu, 15 Apr 2010)
| 20 lines Merged revisions 257467 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r257467 | tilghman | 2010-04-15 15:24:50 -0500 (Thu, 15 Apr 2010)
| 13 lines Don't recreate peer, when responding to a repeated
deregistration attempt. When a reply to a deregistration is lost
in transmit, the client retries the deregistration. Previously,
this would cause a realtime/autocreate peer to be loaded back
into memory, after it had already been correctly purged. Instead,
we just want to resend the reply without loading the peer.
(closes issue #16908) Reported by: kkm Patches:
20100412__issue16908.diff.txt uploaded by tilghman (license 14)
Tested by: kkm ........ ................
2010-04-15 19:42 +0000 [r257344-257428] Leif Madsen <lmadsen@digium.com>
* doc/backtrace.txt: Merged revisions 257427 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r257427 | lmadsen | 2010-04-15 14:41:05 -0500 (Thu, 15 Apr 2010)
| 21 lines Merged revisions 257426 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r257426 | lmadsen | 2010-04-15 14:40:33 -0500 (Thu, 15 Apr 2010)
| 13 lines Update backtrace.txt documentation. Update the
backtrace.txt documentation so it conforms to the same layout as
other documents we've been working on recently. Additionally, add
a bunch of new information about gathering backtraces for crashes
and deadlocks, along with ways of verifying your file before
uploading it. Create a couple of one line commands for people to
generate the files we need. (closes issue #17190) Reported by:
lmadsen Patches: backtrace.txt.patch-2 uploaded by lmadsen
(license 10) Tested by: lmadsen, pabelanger ........
................
* doc/backtrace.txt: Merged revisions 257343 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r257343 | lmadsen | 2010-04-15 08:44:38 -0500 (Thu, 15 Apr 2010)
| 9 lines Merged revisions 257342 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r257342 | lmadsen | 2010-04-15 08:41:45 -0500 (Thu, 15 Apr 2010)
| 1 line Update address of the bug tracker. ........
................
2010-04-14 23:00 +0000 [r257265] Tilghman Lesher <tlesher@digium.com>
* configs/features.conf.sample, /, main/features.c: Merged
revisions 257262 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r257262 |
tilghman | 2010-04-14 17:57:35 -0500 (Wed, 14 Apr 2010) | 15
lines Yet another issue where the conversion of the application
delimiter to comma caused an issue. Application arguments within
the feature map could possibly contain a comma, which conflicts
with the syntax of the features.conf configuration file. This
patch allows the argument to be wrapped in parentheses or quoted,
to allow the application arguments to be interpreted as a single
configuration parameter. (closes issue #16646) Reported by:
pinga-fogo Patches: 20100414__issue16646.diff.txt uploaded by
tilghman (license 14) Tested by: tilghman Review:
https://reviewboard.asterisk.org/r/547/ ........
2010-04-13 19:20 +0000 [r257210] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_sip.c: Merged revisions 257191 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r257191 |
tilghman | 2010-04-13 14:17:48 -0500 (Tue, 13 Apr 2010) | 10
lines Also unref the pvt when we delete the provisional keepalive
job. (closes issue #16774) Reported by: kowalma Patches:
20100315__issue16774.diff.txt uploaded by tilghman (license 14)
Tested by: falves11, jamicque Review:
https://reviewboard.asterisk.org/r/591/ ........
2010-04-13 18:43 +0000 [r257184] Matthew Nicholson <mnicholson@digium.com>
* main/manager.c, /, configs/manager.conf.sample: Merged revisions
257146 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r257146 | mnicholson | 2010-04-13 13:10:30 -0500 (Tue, 13 Apr
2010) | 16 lines Merged revisions 257070 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r257070 | mnicholson | 2010-04-13 11:46:30 -0500 (Tue, 13 Apr
2010) | 9 lines Add an option to restore past broken behavor of
the Events manager action Before r238915, certain values for the
EventMask parameter of the Events action would result in no
response being returned. This patch adds an option to restore
that broken behavior. Also while fixing this bug I discovered
that passing an empty EventMasks parameter would also result in
no response being returned, this has been fixed as well while
being preserved when the broken behavior is requested. (closes
issue #17023) Reported by: nblasgen Review:
https://reviewboard.asterisk.org/r/602/ ........ ................
2010-04-13 16:38 +0000 [r257068] Tilghman Lesher <tlesher@digium.com>
* cdr/cdr_sqlite3_custom.c, /: Merged revisions 257065 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r257065 | tilghman | 2010-04-13 11:33:21 -0500 (Tue, 13 Apr 2010)
| 8 lines Ensure that we can have commas within cdr values.
(closes issue #17001) Reported by: snuffy Patches:
20100412__issue17001.diff.txt uploaded by tilghman (license 14)
Tested by: snuffy ........
2010-04-12 17:30 +0000 [r256822-256902] Leif Madsen <lmadsen@digium.com>
* doc/HOWTO_collect_debug_information.txt (added): Merged revisions
256901 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r256901 | lmadsen | 2010-04-12 12:29:53 -0500 (Mon, 12 Apr 2010)
| 23 lines Merged revisions 256900 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r256900 | lmadsen | 2010-04-12 12:29:26 -0500 (Mon, 12 Apr 2010)
| 15 lines Add How-To document on collecting debugging info for
issues.asterisk.org Paul Belanger has been helping a lot with bug
tracking recently and created this document that we can now point
to when additional debugging information is required. This
document will help those filing issues to know how to get the
information required when filing their issues. This will make
things easier on the developers. Initial text and changes by
pabelanger. Tweaks and editing by myself. (closes issue #17159)
Reported by: pabelanger Patches:
HOWTO_collect_debug_information.txt.patch uploaded by lmadsen
(license 10) Tested by: tzafrir, pabelanger, lmadsen ........
................
* apps/app_voicemail.c, /: Merged revisions 256860 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r256860 | lmadsen | 2010-04-12 11:16:43 -0500 (Mon, 12 Apr 2010)
| 3 lines Remove silly debug message that is not useful. (issue
#17159) ........
* /, main/logger.c: Merged revisions 256821 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r256821 |
lmadsen | 2010-04-12 09:39:37 -0500 (Mon, 12 Apr 2010) | 8 lines
CLI command logger set level auto complete. A simple patch to
enable auto tab complete. (closes issue #17152) Reported by:
pabelanger Patches: 0017152.patch uploaded by pabelanger (license
224) ........
2010-04-08 22:03 +0000 [r256483] Tilghman Lesher <tlesher@digium.com>
* main/app.c: Backport /proc/%d/fd method of closing file
descriptors to 1.6.2.
2010-04-06 19:40 +0000 [r256373] Tilghman Lesher <tlesher@digium.com>
* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
include/asterisk/lock.h: Merged revisions 256370 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r256370 | tilghman | 2010-04-06 14:28:42 -0500 (Tue, 06 Apr 2010)
| 2 lines Mac OS X does not support comparing a mutex to its
initializer. Create a test for this. ........
2010-04-06 18:53 +0000 [r256268-256368] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: CallerID channel DAHDI port FXS are empty
after the first call. The bug is exposed if MFC/R2 support is
built into asterisk (i.e., openr2.h is present in the include
path). Code that unconditionally clears the CallerID name and
number is included. Also fixed a malformed if test in mkintf()
added by issue 15883. Converted the if statement to a switch
statement for clarity. Regression of the issue 15883 fix. (closes
issue #16968) Reported by: grecco Patches: issue16968.patch
uploaded by rmudgett (license 664) (closes issue #16747) Reported
by: viniciusfontes
* channels/chan_dahdi.c, /: Merged revisions 256265 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r256265 | rmudgett | 2010-04-05 19:39:44 -0500
(Mon, 05 Apr 2010) | 12 lines Merged revisions 256225 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r256225 | rmudgett | 2010-04-05 19:10:16 -0500 (Mon, 05 Apr 2010)
| 5 lines DAHDI/PRI call to pri_channel_bridge() not protected by
PRI lock. SWP-1231 ABE-2163 ........ ................
2010-05-03 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.7 Released
2010-04-29 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.7-rc3 Released
2010-04-29 10:31 +0000 [r260053] David Vossel <dvossel@digium.com>
* include/asterisk/audiohook.h, main/audiohook.c: Fixes crash in
audiohook_write_list. (closes issue 0017052) Reported by: dvossel
Tested by: dvossel. (closes issue 0016196) Reported by: atis.
Review: https://reviewboard.asterisk.org/r/623/
2010-04-28 10:31 +0000 [r259899] David Vossel <dvossel@digium.com>
* channels/chan_local.c, main/channel.c: Resolves deadlocks in
chan_local. (closes issue 0017185) Reported by: schmoozecom
Patches: issue_17185_v1.diff uploaded by dvossel (license 671)
issue_17185_v2.diff uploaded by dvossel (license 671) Tested
by: schmoozecom, GameGamer43
Review: https://reviewboard.asterisk.org/r/631/
2010-04-13 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.7-rc2 Released
2010-04-13 [r257210] Tilghman Lesher <tlesher@digium.com>
Also unref the pvt when we delete the provisional keepalive job.
(closes issue #16774)
Reported by: kowalma
Patches:
20100315__issue16774.diff.txt uploaded by tilghman (license 14)
Tested by: falves11, jamicque
Review: https://reviewboard.asterisk.org/r/591/
2010-04-05 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.7-rc1 Released
2010-04-05 15:15 +0000 [r256162] Leif Madsen <lmadsen@digium.com>
* doc/tex/localchannel.tex, /: Merged revisions 256161 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r256161 | lmadsen | 2010-04-05 10:14:53 -0500 (Mon, 05 Apr 2010)
| 1 line Fix for localchannel.tex to allow PDFs to be generated
again. ........
2010-04-02 23:56 +0000 [r256013-256020] Russell Bryant <russell@digium.com>
* /, apps/app_meetme.c: Merged revisions 256019 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r256019 |
russell | 2010-04-02 18:55:57 -0500 (Fri, 02 Apr 2010) | 10 lines
Export MEETMEBOOKID and fix pin-less conferences with realtime
conferences (closes issue #16866) Reported by: DEA Patches:
rt-meetme-options.txt uploaded by DEA (license 3) Tested by: DEA
Review: https://reviewboard.asterisk.org/r/582/ ........
* channels/chan_local.c, /: Merged revisions 256015 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r256015 | russell | 2010-04-02 18:46:45 -0500
(Fri, 02 Apr 2010) | 16 lines Merged revisions 256014 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r256014 | russell | 2010-04-02 18:45:56 -0500 (Fri, 02 Apr 2010)
| 9 lines Resolve a deadlock that occurs due to a pointless call
to ast_bridged_channel() (closes issue #16840) Reported by:
bzing2 Patches: patch.txt uploaded by bzing2 (license 902)
issue_16840.rev1.diff uploaded by russell (license 2) Tested by:
bzing2, russell ........ ................
* main/channel.c, /: Merged revisions 256010 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r256010 | russell | 2010-04-02 18:30:58 -0500 (Fri, 02 Apr 2010)
| 9 lines Merged revisions 256009 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r256009 | russell | 2010-04-02 18:30:15 -0500 (Fri, 02 Apr 2010)
| 2 lines Remove extremely verbose debug message. ........
................
2010-04-02 20:20 +0000 [r255955] Tilghman Lesher <tlesher@digium.com>
* main/asterisk.c, /: Merged revisions 255952 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r255952 |
tilghman | 2010-04-02 15:19:01 -0500 (Fri, 02 Apr 2010) | 8 lines
Pass the PID of the Asterisk process, not the PID of the canary.
(closes issue #17065) Reported by: globalnetinc Patches:
astcanary.patch uploaded by makoto (license 38) Tested by: frawd,
globalnetinc ........
2010-04-01 18:21 +0000 [r255676-255816] Tilghman Lesher <tlesher@digium.com>
* /, include/asterisk/lock.h: Merged revisions 255796 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r255796 | tilghman | 2010-04-01 13:16:37 -0500 (Thu, 01 Apr 2010)
| 7 lines Fix DEBUG_THREADS build on Darwin. (closes issue
#16828) Reported by: oej Patches: 20100331__issue16828.diff.txt
uploaded by tilghman (license 14) ........
* apps/app_voicemail.c, /: Recorded merge of revisions 255592 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r255592 | tilghman | 2010-03-31 14:13:02 -0500
(Wed, 31 Mar 2010) | 22 lines Recorded merge of revisions 255591
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r255591 | tilghman | 2010-03-31 14:09:46 -0500 (Wed, 31 Mar 2010)
| 15 lines Ensure line terminators in email are consistent. Fixes
an issue with certain Mail Transport Agents, where attachments
are not interpreted correctly. (closes issue #16557) Reported by:
jcovert Patches: 20100308__issue16557__1.4.diff.txt uploaded by
tilghman (license 14) 20100308__issue16557__1.6.0.diff.txt
uploaded by tilghman (license 14)
20100308__issue16557__trunk.diff.txt uploaded by tilghman
(license 14) Tested by: ebroad, zktech Reviewboard:
https://reviewboard.asterisk.org/r/544/ ........ ................
2010-03-31 17:49 +0000 [r255505] Leif Madsen <lmadsen@digium.com>
* configs/sip.conf.sample, apps/app_dial.c: Merged revisions 255504
via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r255504 | lmadsen | 2010-03-31 12:48:09 -0500 (Wed, 31
Mar 2010) | 5 lines Add documentation clarifying when 't' and 'T'
can be used. (closes issue #17021) Reported by: kovzol Tested by:
lmadsen, kovzol, davidw, ebroad ........
2010-03-30 20:58 +0000 [r255326-255413] Russell Bryant <russell@digium.com>
* /, channels/chan_h323.c: Merged revisions 255410 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r255410 | russell | 2010-03-30 15:56:26 -0500
(Tue, 30 Mar 2010) | 9 lines Merged revisions 255409 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r255409 | russell | 2010-03-30 15:56:00 -0500 (Tue, 30
Mar 2010) | 2 lines Don't kill Asterisk if the H323 listener does
not start. ........ ................
* /, pbx/pbx_dundi.c: Merged revisions 255323 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r255323 | russell | 2010-03-30 11:07:49 -0500 (Tue, 30 Mar 2010)
| 9 lines Merged revisions 255322 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r255322 | russell | 2010-03-30 11:06:06 -0500 (Tue, 30 Mar 2010)
| 2 lines Don't make Asterisk not start if pbx_dundi fails to
initialize. ........ ................
2010-03-26 19:28 +0000 [r255023-255067] Leif Madsen <lmadsen@digium.com>
* configs/sip.conf.sample, /: Merged revisions 255066 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r255066 | lmadsen | 2010-03-26 14:27:56 -0500 (Fri, 26 Mar 2010)
| 6 lines Replace some documentation from 1.6.x back into trunk.
This documentation associated wth tlsbindaddr is still useful so
lets synchronize it between trunk and 1.6.x branches. (issue
#17054) ........
* configs/sip.conf.sample, /: Merged revisions 255021 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r255021 | lmadsen | 2010-03-26 14:07:38 -0500 (Fri, 26 Mar 2010)
| 8 lines Update confusing documentation for tlsbindaddr. Update
some confusing documentation for the tlsbindaddr option in
sip.conf.sample. Point at a link instead which has better
documentation. (closes issue #17054) Reported by: klaus3000
........
2010-03-25 20:43 +0000 [r254770-254805] Jason Parker <jparker@digium.com>
* utils/Makefile, /: Merged revisions 254802 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r254802 | qwell | 2010-03-25 15:41:49 -0500 (Thu, 25 Mar 2010) |
9 lines Merged revisions 254800 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r254800 | qwell | 2010-03-25 15:41:15 -0500 (Thu, 25 Mar 2010) |
1 line Don't remove local copies of utils in uninstall. ........
................
* main/astobj2.c, include/asterisk/astobj2.h: Fix DEBUG_THREADS
issue with out-of-tree modules. Take 2, without ABI breakage this
time. Review: https://reviewboard.asterisk.org/r/588/
2010-03-25 20:09 +0000 [r254721] Russell Bryant <russell@digium.com>
* channels/chan_usbradio.c, /: Merged revisions 254718 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r254718 | russell | 2010-03-25 15:08:40 -0500 (Thu, 25 Mar 2010)
| 2 lines chan_usbradio depends on alsa. ........
2010-03-25 17:47 +0000 [r254556] Mark Michelson <mmichelson@digium.com>
* include/asterisk/acl.h, /: Merged revisions 254553 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r254553 | mmichelson | 2010-03-25 12:42:36 -0500
(Thu, 25 Mar 2010) | 11 lines Merged revisions 254552 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r254552 | mmichelson | 2010-03-25 12:33:35 -0500 (Thu, 25 Mar
2010) | 5 lines Add doxygen for acl.h Review:
https://reviewboard.asterisk.org/r/528 ........ ................
2010-03-25 17:21 +0000 [r254548] Sean Bright <sean@malleable.com>
* channels/chan_sip.c: Initialize stream to avoid a compilation
error.
2010-03-25 17:12 +0000 [r254542] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Fix potential crashes from trying to
reference nonexistent RTP streams.
2010-03-25 16:26 +0000 [r254499] Terry Wilson <twilson@digium.com>
* /, main/file.c: Merged revisions 254453 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r254453 | twilson | 2010-03-25 11:03:51 -0500 (Thu, 25 Mar 2010)
| 9 lines Merged revisions 254451 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r254451 | twilson | 2010-03-25 10:57:29 -0500 (Thu, 25 Mar 2010)
| 2 lines Handle new SRCCHANGE control message here too ........
................
2010-03-25 16:22 +0000 [r254482] Mark Michelson <mmichelson@digium.com>
* main/rtp.c, /: Recorded merge of revisions 254454 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r254454 | mmichelson | 2010-03-25 11:04:48 -0500
(Thu, 25 Mar 2010) | 50 lines Recorded merge of revisions 254452
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r254452 | mmichelson | 2010-03-25 10:59:56 -0500 (Thu, 25 Mar
2010) | 44 lines Several fixes regarding RFC2833 DTMF detection.
Here is a copy and paste of the details from my request on
reviewboard that dealt with these changes: Fix 1. The first
change in place is to fix Mantis issue 15811, which deals with a
situation where Asterisk will incorrectly interpret out of order
RFC2833 frames as duplicate DTMF digits. For instance, we would
receive a sequence like: seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3:
DTMF 1 seqno 4: DTMF 1 seqno 6: DTMF 1 (end) seqno 5: DTMF 1
seqno 7: DTMF 1 (end) seqno 8: DTMF 1 (end) Prior to this patch
when we received the frame with seqno 5, we would interpret this
as a new DTMF 1. With this patch, we will check the seqno of the
incoming digit and not process the frame if the seqno is lower
than the last recorded seqno. Note that we do not record the
seqno of the dropped DTMF frame for future processing. While the
above situation is what was designed to be fixed, the patch is
written in such a way that the following would also be fixed too:
seqno 9: DTMF 1 seqno 10: DTMF 1 (end) seqno 11: DTMF 1 (end)
seqno 13: DTMF 2 seqno 12: DTMF 1 (end) seqno 14: DTMF 2 seqno
15: DTMF 2 (end) seqno 16: DTMF 2 (end) seqno 17: DTMF 2 (end) In
this second situation, the beginning of the DTMF 2 arrives before
the final end frame of the DTMF 1. With the patch, seqno 12 is no
processed and thus we properly interpret the DTMF. Fix 2. The
second change in place is to fix an issue like the following:
seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3: DTMF 1 (end) *packet
lost* seqno 4: DTMF 1 (end) *packet lost* seqno 5: DTMF 1 (end)
*packet lost* seqno 6: DTMF 2 When we receive seqno 6, we had
code in place that was supposed to properly end the previously
unended DTMF 1. The problem was that the code was essentially a
no-op. The code would set up an end frame for the DTMF 1 but
would immediately overwrite the frame with the begin for DTMF 2.
I changed process_dtmf_rfc2833() so that instead of returning a
single frame, it is given as an output parameter a list of
frames. Each frame that needs to be returned is appended to this
list. Fix 3. The final change is a minor one where an
AST_CONTROL_SRCCHANGE frame could get lost. If we process a cisco
DTMF or an RFC 3389 frame and no frame was returned, then we
would return &ast_null_frame. The problem is that earlier in the
function, we may have generated an AST_CONTROL_SRCCHANGE frame
and put it in the list of frames we wish to return. This frame
would be lost in such a case. The patch fixes this problem
........ ................
2010-03-25 15:21 +0000 [r254447] Leif Madsen <lmadsen@digium.com>
* /, res/res_agi.c: Merged revisions 254446 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r254446 |
lmadsen | 2010-03-25 10:21:26 -0500 (Thu, 25 Mar 2010) | 9 lines
handle_speechset has 4 arguments. Update code to reflect that
handle_speechset has 4 arguments. (closes issue #17093) Reported
by: gpatri Patches: res_agi.patch uploaded by gpatri (license
1014) Tested by: pabelanger, mmichelson ........
2010-03-24 17:19 +0000 [r254288] Jeff Peeler <jpeeler@digium.com>
* res/res_monitor.c, /: Merged revisions 254277 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r254277 | jpeeler | 2010-03-24 12:15:05 -0500 (Wed, 24 Mar 2010)
| 78 lines Merged revisions 254235 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r254235 | jpeeler | 2010-03-23 19:37:23 -0500 (Tue, 23 Mar 2010)
| 72 lines Ensure that monitor recordings are written to the
correct location (again) This is an extension to 248860. As such
the dialplan test has been extended: ; non absolute path, not
combined exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test)
exten => 5040, n, dial(sip/5001) ; absolute path, not combined
exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2) exten =>
5041, n, dial(sip/5001) ; no path, not combined exten => 5042, 1,
monitor(wav,monitor_test3) exten => 5042, n, dial(sip/5001) ;
combined: changemonitor from non absolute to no path (leaves
tmp/jeff) exten => 5043, 1, monitor(wav,tmp/jeff/monitor_test4,m)
exten => 5043, n, changemonitor(monitor_test5) exten => 5043, n,
dial(sip/5001) ; combined: changemonitor from no path to non
absolute path exten => 5044, 1, monitor(wav,monitor_test6,m)
exten => 5044, n, changemonitor(tmp/jeff/monitor_test7) ; this
wasn't possible before exten => 5044, n, dial(sip/5001) ; non
absolute path, combined exten => 5045, 1,
monitor(wav,tmp/jeff/monitor_test8,m) exten => 5045, n,
dial(sip/5001) ; absolute path, combined exten => 5046, 1,
monitor(wav,/tmp/jeff/monitor_test9,m) exten => 5046, n,
dial(sip/5001) ; no path, combined exten => 5047, 1,
monitor(wav,monitor_test10,m) exten => 5047, n, dial(sip/5001) ;
combined: changemonitor from non absolute to absolute (leaves
tmp/jeff) exten => 5048, 1,
monitor(wav,tmp/jeff/monitor_test11,m) exten => 5048, n,
changemonitor(/tmp/jeff/monitor_test12) exten => 5048, n,
dial(sip/5001) ; combined: changemonitor from absolute to non
absolute (leaves /tmp/jeff) exten => 5049, 1,
monitor(wav,/tmp/jeff/monitor_test13,m) exten => 5049, n,
changemonitor(tmp/jeff/monitor_test14) exten => 5049, n,
dial(sip/5001) ; combined: changemonitor from no path to absolute
exten => 5050, 1, monitor(wav,monitor_test15,m) exten => 5050, n,
changemonitor(/tmp/jeff/monitor_test16) exten => 5050, n,
dial(sip/5001) ; combined: changemonitor from absolute to no path
(leaves /tmp/jeff) exten => 5051, 1,
monitor(wav,/tmp/jeff/monitor_test17,m) exten => 5051, n,
changemonitor(monitor_test18) exten => 5051, n, dial(sip/5001) ;
not combined: changemonitor from non absolute to no path (leaves
tmp/jeff) exten => 5052, 1, monitor(wav,tmp/jeff/monitor_test19)
exten => 5052, n, changemonitor(monitor_test20) exten => 5052, n,
dial(sip/5001) ; not combined: changemonitor from no path to non
absolute exten => 5053, 1, monitor(wav,monitor_test21) exten =>
5053, n, changemonitor(tmp/jeff/monitor_test22) exten => 5053, n,
dial(sip/5001) ; not combined: changemonitor from non absolute to
absolute (leaves tmp/jeff) exten => 5054, 1,
monitor(wav,tmp/jeff/monitor_test23) exten => 5054, n,
changemonitor(/tmp/jeff/monitor_test24) exten => 5054, n,
dial(sip/5001) ; not combined: changemonitor from absolute to non
absolute (leaves /tmp/jeff) exten => 5055, 1,
monitor(wav,/tmp/jeff/monitor_test24) exten => 5055, n,
changemonitor(tmp/jeff/monitor_test25) exten => 5055, n,
dial(sip/5001) ; not combined: changemonitor from no path to
absolute exten => 5056, 1, monitor(wav,monitor_test26) exten =>
5056, n, changemonitor(/tmp/jeff/monitor_test27) exten => 5056,
n, dial(sip/5001) ; not combined: changemonitor from absolute to
no path (leaves /tmp/jeff) exten => 5057, 1,
monitor(wav,/tmp/jeff/monitor_test28) exten => 5057, n,
changemonitor(monitor_test29) exten => 5057, n, dial(sip/5001)
........ ................
2010-03-23 22:05 +0000 [r254131] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* tests/Makefile, /: Merged revisions 254001 via svnmerge from
http://svn.digium.com/svn/asterisk/trunk ........ r254001 |
tzafrir | 2010-03-23 21:19:52 +0200 (Tue, 23 Mar 2010) | 2 lines
Change the name of the category 'TEST' to match the name of the
subdir ........
2010-03-23 21:20 +0000 [r254068] Jeff Peeler <jpeeler@digium.com>
* main/channel.c, /: Merged revisions 254050 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r254050 |
jpeeler | 2010-03-23 16:17:23 -0500 (Tue, 23 Mar 2010) | 14 lines
Exit native bridging early for greater timing accuracy with
warnings This changes native bridging to break one millisecond
early so that the more accurate timeval calculations done in the
generic bridge can be performed using the bridge config.
Currently the time between exiting native bridging slightly late
can sometimes cause a large enough discrepancy for warnings to be
missed. For the record, 1.4 does not attempt to native bridge at
all when warnings are enabled. (closes issue #15815) Reported by:
adomjan Review: https://reviewboard.asterisk.org/r/577/ ........
2010-03-22 19:55 +0000 [r253801] Matthew Nicholson <mnicholson@digium.com>
* /, main/features.c: Merged revisions 253800 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r253800 | mnicholson | 2010-03-22 14:52:52 -0500 (Mon, 22 Mar
2010) | 11 lines Merged revisions 253799 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r253799 | mnicholson | 2010-03-22 14:50:00 -0500 (Mon, 22 Mar
2010) | 4 lines Unconditionally copy the caller's account code to
the called party. (related to issue #16331) ........
................
2010-03-22 19:06 +0000 [r253714-253760] Tilghman Lesher <tlesher@digium.com>
* /, contrib/scripts/dbsep.cgi: Merged revisions 253758 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r253758 | tilghman | 2010-03-22 14:05:27 -0500 (Mon, 22
Mar 2010) | 2 lines Update query should be an UPDATE, not a
SELECT. ........
* /, contrib/scripts/dbsep.cgi: Merged revisions 253755 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r253755 | tilghman | 2010-03-22 13:58:48 -0500 (Mon, 22
Mar 2010) | 4 lines Return the list for later manipulation. This
fixes an issue with the update procedure. Debugging with
mmichelson. ........
* configs/dbsep.conf.sample, /, contrib/scripts/dbsep.cgi: Merged
revisions 253712 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r253712 |
tilghman | 2010-03-22 11:59:35 -0500 (Mon, 22 Mar 2010) | 2 lines
Accomodate equal signs in DSNs and add documentation, based upon
mmichelson's feedback. ........
2010-03-20 17:33 +0000 [r253595-253620] Russell Bryant <russell@digium.com>
* cdr/cdr_pgsql.c, main/stdtime/localtime.c, main/tcptls.c, /,
main/features.c: Merged revisions 253540 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r253540 |
russell | 2010-03-20 07:03:07 -0500 (Sat, 20 Mar 2010) | 2 lines
Resolve more compiler warnings on FreeBSD. ........
* apps/app_followme.c, apps/app_dial.c, /: Merged revisions 253538
via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r253538 | russell | 2010-03-20 06:43:08 -0500 (Sat, 20
Mar 2010) | 2 lines Resolve compiler warnings on FreeBSD.
........
* /, pbx/pbx_dundi.c: Merged revisions 253537 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r253537 |
russell | 2010-03-20 06:39:39 -0500 (Sat, 20 Mar 2010) | 2 lines
Resolve a compiler warning on FreeBSD. ........
* channels/chan_dahdi.c, /: Merged revisions 253536 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r253536 | russell | 2010-03-20 06:33:30 -0500 (Sat, 20 Mar 2010)
| 4 lines Use SHRT_MAX instead of MAXSHORT. These changes fix
build issues I had with this module on FreeBSD. ........
2010-03-19 08:05 +0000 [r253492] Alec L Davis <sivad.a@paradise.net.nz>
* main/astobj2.c, /: Merged revisions 253490 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r253490 |
alecdavis | 2010-03-19 20:37:00 +1300 (Fri, 19 Mar 2010) | 19
lines prevent segfault if bad magic number is encountered.
internal_ao2_ref uses INTERNAL_OBJ which mzy report 'bad magic
number', but internal_ao2_ref continues on, causing segfault.
Although AO2_MAGIC number is checked by INTERNAL_OBJ before
internal_ao2_ref is called, A02_MAGIC is being destroyed (or a
wrong pointer) by the time internal_ao2_ref uses INTERNAL_OBJ.
internal_ao2_ref now returns -1 if INTERNAL_OBJ encouters a bad
magic number. (issue #17037) Reported by: alecdavis Patches:
bug17037.diff.txt uploaded by alecdavis (license 585) Tested by:
alecdavis ........
2010-03-18 17:54 +0000 [r253257-253346] Leif Madsen <lmadsen@digium.com>
* /, apps/app_userevent.c: Merged revisions 253345 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r253345 | lmadsen | 2010-03-18 12:52:35 -0500 (Thu, 18 Mar 2010)
| 7 lines Change usage of pipe to comma in UserEvent docs. Change
the example usage of pipe as a separator to comma in the
UserEvent documentation. (closes issue #16961) Reported by:
jlpedrosa ........
* doc/tex/localchannel.tex: Merged revisions 253256 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r253256 | lmadsen | 2010-03-18 10:46:52 -0500 (Thu, 18 Mar 2010)
| 9 lines Update to new Local channel documentation. Add same
changes as commit to 1.4, but convert to TeX. (issue #16963)
Reported by: kobaz Patches: localchannel-2.txt uploaded by kobaz
(license 834) ........
2010-03-17 16:25 +0000 [r253158] Terry Wilson <twilson@digium.com>
* main/rtp.c, channels/chan_skinny.c, channels/chan_h323.c,
channels/chan_mgcp.c, channels/chan_sip.c,
include/asterisk/rtp.h: Revert API change in release branches
This re-renames ast_rtp_update_source to ast_rtp_new_source
2010-03-17 00:41 +0000 [r253029-253033] Leif Madsen <lmadsen@digium.com>
* main/xmldoc.c, /: Merged revisions 253032 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r253032 |
lmadsen | 2010-03-16 19:40:51 -0500 (Tue, 16 Mar 2010) | 1 line
Fix a typo. ........
* configs/say.conf.sample, /: Merged revisions 253028 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r253028 | lmadsen | 2010-03-16 19:29:06 -0500
(Tue, 16 Mar 2010) | 13 lines Merged revisions 253018 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r253018 | lmadsen | 2010-03-16 19:26:19 -0500 (Tue, 16 Mar 2010)
| 6 lines Add french snipset to say.conf. Add the french snipset
to say.conf. (Closes issue #15799) ........ ................
2010-03-16 23:54 +0000 [r252978] Tilghman Lesher <tlesher@digium.com>
* apps/app_stack.c, /: Merged revisions 252976 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r252976 |
tilghman | 2010-03-16 18:49:35 -0500 (Tue, 16 Mar 2010) | 8 lines
Mask out previous arguments on each nested invocation of Gosub.
(closes issue #16758) Reported by: wdoekes Patches:
20100316__issue16758.diff.txt uploaded by tilghman (license 14)
Review: https://reviewboard.asterisk.org/r/561/ ........
2010-03-16 19:38 +0000 [r252850] Sean Bright <sean@malleable.com>
* res/res_clialiases.c, /: Merged revisions 252848 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r252848 | seanbright | 2010-03-16 15:36:24 -0400 (Tue, 16 Mar
2010) | 10 lines Include an extra newline after "Aliased CLI
command" to get back the prompt. The other issue mentioned in
this bug will be more difficult to resolve since we have no idea
(right now) of knowing if the command that is aliased has been
installed yet. (issue #16978) Reported by: jw-asterisk Tested by:
seanbright ........
2010-03-16 19:02 +0000 [r252770] Russell Bryant <russell@digium.com>
* utils/Makefile, /: Merged revisions 252767 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r252767 | russell | 2010-03-16 14:01:04 -0500 (Tue, 16 Mar 2010)
| 13 lines Merged revisions 252766 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r252766 | russell | 2010-03-16 14:00:43 -0500 (Tue, 16 Mar 2010)
| 6 lines Don't treat warnings as errors for muted. muted
supports OS X, but uses functions marked as deprecated in 10.6.
However, the functions are still supported, so just ignore the
warnings for now and allow the build to proceed. ........
................
2010-03-16 18:49 +0000 [r252763] Leif Madsen <lmadsen@digium.com>
* configs/extensions.ael.sample, /: Merged revisions 252762 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r252762 | lmadsen | 2010-03-16 13:48:22 -0500
(Tue, 16 Mar 2010) | 15 lines Merged revisions 252761 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r252761 | lmadsen | 2010-03-16 13:46:20 -0500 (Tue, 16 Mar 2010)
| 7 lines Additional extensions.ael global variable fixes. Fixing
up a couple more overlapping global variable namespaces shared
with extensions.conf.sample. Also noticed a few of the lines that
were commented out didn't have the closing semi-colon so I added
that as well. (issue #17035) ........ ................
2010-03-15 21:59 +0000 [r252626] Sean Bright <sean@malleable.com>
* /, apps/app_meetme.c: Merged revisions 252623 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r252623 |
seanbright | 2010-03-15 17:55:44 -0400 (Mon, 15 Mar 2010) | 4
lines Resolve a crash in SLATrunk when the specified trunk
doesn't exist. Reported by philipp64 in #asterisk-dev. ........
2010-03-15 21:54 +0000 [r252622] Tilghman Lesher <tlesher@digium.com>
* contrib/init.d/org.asterisk.asterisk.plist, /: Merged revisions
252619 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r252619 | tilghman | 2010-03-15 16:51:55 -0500 (Mon, 15 Mar 2010)
| 9 lines Merged revisions 252617 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r252617 | tilghman | 2010-03-15 16:43:14 -0500 (Mon, 15 Mar 2010)
| 2 lines Uh, yeah. Umask. I'm stupid. ........ ................
2010-03-15 20:53 +0000 [r252535] Leif Madsen <lmadsen@digium.com>
* configs/extensions.ael.sample: Merged revisions 252534 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r252534 | lmadsen | 2010-03-15 15:52:32 -0500
(Mon, 15 Mar 2010) | 15 lines Merged revisions 252533 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r252533 | lmadsen | 2010-03-15 15:48:56 -0500 (Mon, 15 Mar 2010)
| 7 lines Update extensions.ael file to not overlap
extensions.conf. Updated the extensions.ael file so the global
variables don't overlap those that we have in extensions.conf
(sample files). This way unexpected things won't happed hopefully
if both pbx_ael and res_config are loaded. (closes issue #17035)
Reported by: pprindeville ........ ................
2010-03-15 05:04 +0000 [r252365-252444] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_sip.c: Merged revisions 252442 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r252442 |
tilghman | 2010-03-14 23:25:35 -0500 (Sun, 14 Mar 2010) | 7 lines
THIS IS NOT PYTHON. Indentation doesn't matter, only braces do.
(closes issue #17025) Reported by: smurfix Patches: sip.patch
uploaded by smurfix (license 547) ........
* main/asterisk.c, Makefile,
contrib/init.d/org.asterisk.asterisk.plist (added), /: Merged
revisions 252362 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r252362 | tilghman | 2010-03-14 20:37:04 -0500 (Sun, 14 Mar 2010)
| 11 lines Merged revisions 252361 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r252361 | tilghman | 2010-03-14 20:33:50 -0500 (Sun, 14 Mar 2010)
| 4 lines Launch Asterisk on Mac OS X with launchd. Reviewboard:
https://reviewboard.asterisk.org/r/551/ ........ ................
2010-03-14 17:48 +0000 [r252317] Sean Bright <sean@malleable.com>
* cdr/cdr_sqlite3_custom.c, /: Merged revisions 252314 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r252314 | seanbright | 2010-03-14 13:43:46 -0400 (Sun, 14 Mar
2010) | 8 lines Fix building CDR and CEL SQLite3 modules. They
added a sqlite3_log() function which was conflicting with our
function names. (closes issue #17017) Reported by: alephlg
........
2010-03-13 00:32 +0000 [r252137-252178] Terry Wilson <twilson@digium.com>
* main/rtp.c: Remove unusued field
* configs/sip.conf.sample, include/asterisk/frame.h, main/rtp.c,
channels/chan_mgcp.c, main/channel.c, /, channels/chan_sip.c,
channels/chan_skinny.c, include/asterisk/rtp.h,
channels/chan_h323.c: Merged revisions 252089 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r252089 |
twilson | 2010-03-12 16:04:51 -0600 (Fri, 12 Mar 2010) | 20 lines
Only change the RTP ssrc when we see that it has changed This
change basically reverts the change reviewed in
https://reviewboard.asterisk.org/r/374/ and instead limits the
updating of the RTP synchronization source to only those times
when we detect that the other side of the conversation has
changed the ssrc. The problem is that SRCUPDATE control frames
are sent many times where we don't want a new ssrc, including
whenever Asterisk has to send DTMF in a normal bridge. This is
also not the first time that this mistake has been made. The
initial implementation of the ast_rtp_new_source function also
changed the ssrc--and then it was removed because of this same
issue. Then, we put it back in again to fix a different issue.
This patch attempts to only change the ssrc when we see that the
other side of the conversation has changed the ssrc. It also
renames some functions to make their purpose more clear. Review:
https://reviewboard.asterisk.org/r/540/ ........
2010-03-12 22:05 +0000 [r252090] Moises Silva <moises.silva@gmail.com>
* channels/chan_dahdi.c, /: Merged revisions 252088 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r252088 | moy | 2010-03-12 16:57:40 -0500 (Fri, 12 Mar 2010) | 1
line add missing mfcr2_skip_category setting ........
2010-03-12 19:50 +0000 [r251994] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c, /: Merged revisions 251989 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r251989 | tilghman | 2010-03-12 13:43:23 -0600 (Fri, 12 Mar 2010)
| 8 lines Don't override a user option with the global option.
(closes issue #16849) Reported by: ip-rob Patches:
20100311__issue16849.diff.txt uploaded by tilghman (license 14)
Tested by: ip-rob ........
2010-03-12 19:49 +0000 [r251991] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 251946 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r251946 | rmudgett | 2010-03-12 13:05:40 -0600 (Fri, 12 Mar 2010)
| 1 line Doxegen this chan_dahdi lock. ........
2010-03-11 21:08 +0000 [r251879-251887] Tilghman Lesher <tlesher@digium.com>
* apps/app_exec.c, /: Merged revisions 251884 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r251884 |
tilghman | 2010-03-11 15:07:07 -0600 (Thu, 11 Mar 2010) | 8 lines
Because ExecIf needs to reprocess arguments, it's best if we
don't remove quotes during parsing. (closes issue #16905)
Reported by: ip-rob Patches: 20100303__issue16905.diff.txt
uploaded by tilghman (license 14) Tested by: ip-rob ........
* apps/app_system.c, /: Merged revisions 251877 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r251877 |
tilghman | 2010-03-11 14:25:02 -0600 (Thu, 11 Mar 2010) | 8 lines
If the argument to the system application is quoted, ensure we
remove the quotes before trying to execute. (closes issue #16842)
Reported by: ip-rob Patches: 20100310__issue16842.diff.txt
uploaded by tilghman (license 14) Tested by: ip-rob ........
2010-03-11 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.6 released
2010-03-05 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.6-rc2 released
2010-03-05 Tilghman Lesher <tlesher@digium.com>
* /, apps/app_voicemail.c: Merged revisions 250913 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r250913 | tilghman
| 2010-03-04 22:37:36 -0600 (Thu, 04 Mar 2010) | 7 lines Missing quote in
ODBC query. (closes issue #16953) Reported by: elguero Patches:
app_voicemail-odbc-syntax-fix.diff uploaded by elguero (license 37)
........
2010-03-04 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.6-rc1 released
2010-03-03 21:24 +0000 [r250610] Leif Madsen <lmadsen@digium.com>
* doc/tex/localchannel.tex, /: Merged revisions 250609 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r250609 | lmadsen | 2010-03-03 16:22:55 -0500 (Wed, 03 Mar 2010)
| 11 lines Update existing Local channel documentation. A
complete re-write of the Local channel documentation has been
performed, with the existing information from localchannel.txt
and localchannel.tex merged in. (closes issue #16637) Reported
by: kobaz Patches: localchannel.tex uploaded by lmadsen (license
10) localchannel.txt uploaded by lmadsen (license 10) Tested by:
lmadsen, jsmith, mmichelson ........
2010-03-03 19:13 +0000 [r250484] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 250481 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r250481 | jpeeler | 2010-03-03 13:06:06 -0600
(Wed, 03 Mar 2010) | 22 lines Merged revisions 250480 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r250480 | jpeeler | 2010-03-03 13:04:11 -0600 (Wed, 03 Mar 2010)
| 15 lines Make sure to clear red alarm after polarity reversal.
From the issue: The automatic overnight line tests (or manual
ones) used on UK (BT) lines causes a red alarm on a dahdi /
TDM400P connected channel. This is because the line uses voltage
tests (battery loss) and polarity reversal. The polarity reversal
causes chan_dahdi to initiate v23 CallerID processing but during
this the event DAHDI_EVENT_NOALARM is ignored so that the alarm
is never cleared. (closes issue #14163) Reported by: jedi98
Patches: chan_dahdi-1.4-inalarm.diff uploaded by jedi98 (license
653) Tested by: mattbrown, Chainsaw, mikeeccleston ........
................
2010-03-03 18:05 +0000 [r250253-250396] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c, /: Merged revisions 250395 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r250395 | dvossel | 2010-03-03 12:03:19 -0600
(Wed, 03 Mar 2010) | 22 lines Merged revisions 250394 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r250394 | dvossel | 2010-03-03 12:02:27 -0600 (Wed, 03 Mar 2010)
| 16 lines fixes problem with duplicate TXREQ packets When
Asterisk receives an IAX2 TXREQ packet, try_transfer() will call
store_by_transfercallno() to link the chan_iax2_pvt struct into
iax_transfercallno_pvts. If a duplicate TXREQ packet is received
for the same call, the pvt struct will be linked into
iax_transfercallno_pvts multiple times. This patch fixes this.
Thanks rain for debugging this and providing a patch! (closes
issue #16904) Reported by: rain Patches:
iax2-double-txreq-fix.diff uploaded by rain (license 327) Tested
by: rain, dvossel ........ ................
* /, channels/chan_sip.c: Merged revisions 250246 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r250246 |
dvossel | 2010-03-02 18:18:28 -0600 (Tue, 02 Mar 2010) | 2 lines
fixes signed to unsigned int comparision issue for FaxMaxDatagram
value. ........
2010-03-02 21:10 +0000 [r249953-250052] Leif Madsen <lmadsen@digium.com>
* doc/tex/imapstorage.tex, /: Merged revisions 250051 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r250051 | lmadsen | 2010-03-02 16:09:27 -0500 (Tue, 02 Mar 2010)
| 8 lines Update IMAP documentation. Update the IMAP
documentation to make it clear that storing voicemails in the
same folder as a large number of emails could potentially cause
significant slow downs when writing or retrieving voicemails.
(issue #16704) Reported by: TimeHider Tested by: lmadsen,
TimeHider ........
* configs/cdr.conf.sample: Merged revisions 250045 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r250045 | lmadsen | 2010-03-02 15:52:19 -0500
(Tue, 02 Mar 2010) | 15 lines Merged revisions 250043 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r250043 | lmadsen | 2010-03-02 15:51:35 -0500 (Tue, 02 Mar 2010)
| 7 lines Update documentation to clarify purpose of unanswered
option. (closes issue #16267) Reported by: elsto Patches:
cdr.conf.sample.patch.txt uploaded by lmadsen (license 10) Tested
by: davidw, elsto ........ ................
* doc/tex/configuration.tex, /: Merged revisions 250037 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r250037 | lmadsen | 2010-03-02 15:36:10 -0500 (Tue, 02
Mar 2010) | 4 lines Update documentation to not imply we support
overriding options. (closes issue #16855) Reported by: davidw
........
* apps/app_directory.c, /: Merged revisions 249950 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r249950 | lmadsen | 2010-03-02 14:49:48 -0500 (Tue, 02 Mar 2010)
| 4 lines Fix literal values wrapped in documentation. (closes
issue #16145) Reported by: tilghman ........
2010-03-02 19:50 +0000 [r249952] Alec L Davis <sivad.a@paradise.net.nz>
* UPGRADE-1.6.txt, main/editline/makelist.in, apps/app_echo.c,
UPGRADE.txt: revert ability to exit echo app caused a regression,
as only supported VOICE, not VIDEO etc. (issue #16880)
2010-03-02 19:26 +0000 [r249916-249933] Leif Madsen <lmadsen@digium.com>
* /, main/features.c: Merged revisions 249925 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r249925 |
lmadsen | 2010-03-02 14:24:43 -0500 (Tue, 02 Mar 2010) | 6 lines
Add missing description of the PARKINGLOT variable in XML
documentation. (closes issue #16743) Reported by: snuffy Patches:
parkingdoc.diff uploaded by snuffy (license 35) ........
* /, pbx/pbx_dundi.c: Merged revisions 249912 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r249912 |
lmadsen | 2010-03-02 14:21:19 -0500 (Tue, 02 Mar 2010) | 6 lines
Convert some DUNDI functions to XML documentation. (closes issue
#16798) Reported by: snuffy Patches: xml_dundi.diff uploaded by
snuffy (license 35) ........
2010-03-02 19:12 +0000 [r249895] David Vossel <dvossel@digium.com>
* channels/chan_console.c, channels/chan_gtalk.c,
channels/chan_oss.c, channels/misdn_config.c,
include/asterisk/abstract_jb.h, configs/alsa.conf.sample,
channels/chan_jingle.c, channels/chan_usbradio.c,
channels/chan_dahdi.c, channels/chan_skinny.c,
configs/mgcp.conf.sample, main/abstract_jb.c,
channels/chan_h323.c, channels/chan_alsa.c,
configs/sip.conf.sample, channels/chan_mgcp.c,
channels/chan_unistim.c, configs/console.conf.sample,
configs/chan_dahdi.conf.sample, channels/chan_local.c,
configs/oss.conf.sample, channels/chan_sip.c, /,
configs/usbradio.conf.sample, configs/misdn.conf.sample: Merged
revisions 249893 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r249893 |
dvossel | 2010-03-02 13:08:38 -0600 (Tue, 02 Mar 2010) | 11 lines
fixes adaptive jitterbuffer configuration When configuring the
adaptive jitterbuffer, the target_extra value not only could not
be set from the configuration, but was not even being set to its
proper default. This value is required in order for the adaptive
jitterbuffer to work correctly. To resolve this a config option
has been added to expose this value to the conf files, and a
default value is provided when no config specific value is
present. ........
2010-03-02 19:09 +0000 [r249894] Leif Madsen <lmadsen@digium.com>
* /, apps/app_confbridge.c: Merged revisions 249892 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r249892 | lmadsen | 2010-03-02 14:02:56 -0500 (Tue, 02 Mar 2010)
| 1 line Fix several XML documentation validate errors. ........
2010-03-02 09:05 +0000 [r249844] Alec L Davis <sivad.a@paradise.net.nz>
* apps/app_echo.c: fixes ability to exit echo app when called from
a ISDN channel, null frames prevent '#' exit. Now only echo back
VOICE and DTMF frames (issue #16880) Reported by: alecdavis
Patches: echo_exit_1-6-1.diff.txt uploaded by alecdavis (license
585) Tested by: alecdavis
2010-03-01 19:40 +0000 [r249675] Sean Bright <sean@malleable.com>
* apps/app_voicemail.c, /: Merged revisions 249672 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r249672 | seanbright | 2010-03-01 14:36:30 -0500
(Mon, 01 Mar 2010) | 18 lines Merged revisions 249671 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r249671 | seanbright | 2010-03-01 14:35:01 -0500 (Mon, 01 Mar
2010) | 11 lines Fix crash in app_voicemail related to message
counting. We were passing a 'struct inprocess **' and treating it
like a 'struct inprocess *' causing a segfault. (closes issue
#16921) Reported by: whardier Patches: 20100301_issue16921.patch
uploaded by seanbright (license 71) Tested by: whardier ........
................
2010-03-01 18:47 +0000 [r249625] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c, /: Merged revisions 249623 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r249623 | tilghman | 2010-03-01 12:36:06 -0600 (Mon, 01 Mar 2010)
| 2 lines Constify a bit of app_voicemail, to make ODBC and IMAP
compile once again. ........
2010-03-01 17:25 +0000 [r249580] Jeff Peeler <jpeeler@digium.com>
* channels/chan_local.c, /: Merged revisions 249538 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r249538 | jpeeler | 2010-03-01 11:11:31 -0600
(Mon, 01 Mar 2010) | 18 lines Merged revisions 249536 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r249536 | jpeeler | 2010-03-01 11:02:03 -0600 (Mon, 01 Mar 2010)
| 11 lines Modify queued frames from local channels to not set
the other side to up In this case, attended transfers were broken
due to ast_feature_request_and_dial detecting the channel being
set to up before the answer frame could be read and therefore
failing to mark the channel as ready. This fix is a regression
fix for 244785, which should continue to work properly as well.
(closes issue #16816) Reported by: jamhed Tested by: jamhed,
corruptor ........ ................
2010-02-28 20:51 +0000 [r249407-249493] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c, /: Merged revisions 249491 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r249491 | tilghman | 2010-02-28 14:50:01 -0600 (Sun, 28 Feb 2010)
| 5 lines Fix unit test that Alec Davis broke. (closes issue
#16927) Reported by: alecdavis ........
* apps/app_voicemail.c, include/asterisk/app.h, /: Merged revisions
249405 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r249405 |
tilghman | 2010-02-28 01:10:22 -0600 (Sun, 28 Feb 2010) | 2 lines
Properly document voicemail API documents. Also fix a crash
reported via the -dev list. ........
2010-02-27 23:04 +0000 [r249321] Alec L Davis <sivad.a@paradise.net.nz>
* channels/chan_dahdi.c: overlap receiving: automatically send CALL
PROCEEDING when dialplan starts Following Q.931 5.2.4 When the
user has determined that sufficient call information has been
received the user shall stop T302 and send CALL PROCEEDING to the
network. Previously timeouts were possible if the dialplan took a
long time to issue any response back to the network. Verified
that our local TELCO also does the same. (issue #16789) Reported
by: alecdavis Patches: overlap_receiving_trunk.diff.txt uploaded
by alecdavis (license 585) Tested by: alecdavis
2010-02-27 14:10 +0000 [r249238] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_iax2.c, /: Merged revisions 249235 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r249235 | kpfleming | 2010-02-27 09:08:35 -0500
(Sat, 27 Feb 2010) | 9 lines Merged revisions 249234 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r249234 | kpfleming | 2010-02-27 09:07:59 -0500 (Sat, 27
Feb 2010) | 1 line add a reference to the now-published IAX2 RFC
........ ................
2010-02-26 18:49 +0000 [r249190] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c, /: Merged revisions 249187 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r249187 | tilghman | 2010-02-26 12:41:57 -0600 (Fri, 26 Feb 2010)
| 18 lines Cleanups to fix bugs in the VM count API functions. -
Urgent voicemails were not attached, because the attachment code
looked in the wrong folder. - Urgent voicemails were sometimes
counted twice when displaying the count of new messages. -
Backends were inconsistent as to which voicemails each API
counted. (closes issue #15654) Reported by: tomo1657 Patches:
20100225__issue15654.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman (closes issue #16448) Reported by: hevad
Review: https://reviewboard.asterisk.org/r/525/ ........
2010-02-26 17:06 +0000 [r249104] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 249101 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r249101 | mmichelson | 2010-02-26 11:04:58 -0600 (Fri, 26 Feb
2010) | 14 lines Merged revisions 249100 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r249100 | mmichelson | 2010-02-26 11:04:29 -0600 (Fri, 26 Feb
2010) | 8 lines For T.38 reINVITEs treat a 606 the same as a 488.
(closes issue #16792) Reported by: vrban Patches: t38_606.patch
uploaded by vrban (license 756) ........ ................
2010-02-25 23:12 +0000 [r248955] Jeff Peeler <jpeeler@digium.com>
* res/res_monitor.c, /: Merged revisions 248952 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r248952 | jpeeler | 2010-02-25 17:09:54 -0600 (Thu, 25 Feb 2010)
| 24 lines Merged revisions 248860 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r248860 | jpeeler | 2010-02-25 15:22:06 -0600 (Thu, 25 Feb 2010)
| 18 lines Ensure that monitor recordings are written to the
correct location (again) This is an extension to 248757. As such
the dialplan test has been extended: exten => 5040, 1,
monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n,
dial(sip/5001) exten => 5041, 1,
monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n,
dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b)
exten => 5042, n, dial(sip/5001) exten => 5043, 1,
monitor(wav,tmp/jeff/monitor_test3,m) exten => 5043, n,
changemonitor(monitor_test4) exten => 5043, n, dial(sip/5001)
exten => 5044, 1, monitor(wav,monitor_test4,m) exten => 5044, n,
changemonitor(tmp/jeff/monitor_test5) ; this looks to fail by
design and emits a warning exten => 5044, n, dial(sip/5001)
........ ................
2010-02-25 22:42 +0000 [r248949] Mark Michelson <mmichelson@digium.com>
* /, main/acl.c: Merged revisions 248946 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r248946 |
mmichelson | 2010-02-25 16:41:48 -0600 (Thu, 25 Feb 2010) | 5
lines Fix incorrect ACL behavior when CIDR notation of "/0" is
used. AST-2010-003 ........
2010-02-25 21:25 +0000 [r248864] Tilghman Lesher <tlesher@digium.com>
* main/asterisk.c, /: Merged revisions 248861 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r248861 | tilghman | 2010-02-25 15:22:39 -0600 (Thu, 25 Feb 2010)
| 22 lines Merged revisions 248859 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r248859 | tilghman | 2010-02-25 15:21:05 -0600 (Thu, 25 Feb 2010)
| 15 lines Some platforms clear /var/run at boot, which makes
connecting a remote console... difficult. Previously, we only
created the default /var/run/asterisk directory at install time.
While we could create it in the init script, that would not work
for those who start asterisk manually from the command line. So
the safest thing to do is to create it as part of the Asterisk
boot process. This also changes the ownership of the directory,
because the pid and ctl files are created after we setuid/setgid.
(closes issue #16802) Reported by: Brian Patches:
20100224__issue16802.diff.txt uploaded by tilghman (license 14)
Tested by: tzafrir ........ ................
2010-02-25 18:52 +0000 [r248797] Jeff Peeler <jpeeler@digium.com>
* res/res_monitor.c, /: Merged revisions 248793 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r248793 | jpeeler | 2010-02-25 12:37:56 -0600 (Thu, 25 Feb 2010)
| 22 lines Merged revisions 248757 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r248757 | jpeeler | 2010-02-25 12:06:54 -0600 (Thu, 25 Feb 2010)
| 15 lines Ensure that monitor recordings are written to the
correct location. Recordings should be placed in the monitor
directory when a non-absolute path is used. Exact dialplan used
for testing: exten => 5040, 1,
monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n,
dial(sip/5001) exten => 5041, 1,
monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n,
dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b)
exten => 5042, n, dial(sip/5001) ABE-2101 ........
................
2010-02-24 21:29 +0000 [r248643] Tilghman Lesher <tlesher@digium.com>
* /, main/logger.c: Merged revisions 248584 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r248584 | tilghman | 2010-02-24 15:17:26 -0600 (Wed, 24 Feb 2010)
| 14 lines Merged revisions 248582 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r248582 | tilghman | 2010-02-24 15:02:18 -0600 (Wed, 24 Feb 2010)
| 7 lines Remove color code sequences from verbose messages that
go to logfiles. (closes issue #16786) Reported by: dodo Patches:
logger2.patch uploaded by dodo (license 989) Tested by: tilghman
........ ................
2010-02-23 16:37 +0000 [r248398] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 248397 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r248397 | dvossel | 2010-02-23 10:34:39 -0600 (Tue, 23 Feb 2010)
| 15 lines Merged revisions 248396 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r248396 | dvossel | 2010-02-23 10:26:05 -0600 (Tue, 23 Feb 2010)
| 9 lines fixes invite with replaces deadlock (closes issue
#16862) Reported by: pwalker Patches: replaces_deadlock_1.4
uploaded by dvossel (license 671) Tested by: pwalker, dvossel
........ ................
2010-02-19 19:07 +0000 [r248011] Tilghman Lesher <tlesher@digium.com>
* channels/chan_console.c, main/loader.c, /: Merged revisions
228798 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r228798 |
tilghman | 2009-11-09 01:37:52 -0600 (Mon, 09 Nov 2009) | 14
lines Fix various problems detected with Valgrind. * chan_console
accessed pvts after deallocation. * The module loader did not
check usecount on shutdown, which led to chan_iax2 reading a
timer that was already unloaded. (closes issue #16062) Reported
by: alexanderheinz Patches: 20091109__issue16062.diff.txt
uploaded by tilghman (license 14) Tested by: tilghman ........
2010-02-19 19:00 +0000 [r248005] Moises Silva <moises.silva@gmail.com>
* channels/chan_dahdi.c, /: Merged revisions 248003 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r248003 | moy | 2010-02-19 13:38:34 -0500 (Fri, 19 Feb 2010) | 1
line mfcr2 issue 0016844 - Fix portability bit fields and make
mfcr2_immediate_accept work again, reported and patched by
korihor ........
2010-02-19 18:45 +0000 [r248004] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c, /: Merged revisions 247914 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r247914 | rmudgett | 2010-02-19 11:33:33 -0600
(Fri, 19 Feb 2010) | 62 lines Merged revisions 247910 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
................ r247910 | rmudgett | 2010-02-19 11:18:49 -0600
(Fri, 19 Feb 2010) | 55 lines Merged revision 247904 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.2-...
.......... r247904 | rmudgett | 2010-02-19 10:49:44 -0600 (Fri,
19 Feb 2010) | 49 lines Make chan_misdn DTMF processing
consistent with other channel technologies. The processing of
DTMF tones on the receiving side of an ISDN channel is
inconsistent with the way it is handled in other channels,
especially DAHDI analog. This causes DTMF tones sent from an ISDN
phone to be doubled at the connected party. We are using the
following 2 options of misdn.conf 1) astdtmf=yes 2) senddtmf=yes
Option one is necessary because the asterisk DSP DTMF detection
is better than mISDN's internal DSP. Not as many false positives.
Option two is necessary to transmit DTMF tones end to end when
mISDN channels are connected to SIP channels with out of band
DTMF for example. The symptom is that DTMF tones sent by an ISDN
phone are doubled on the way through asterisk when two mISDN
channels are connected with a Local channel in between or if it
is bridged to an analog channel. The doubling of DTMF tones is
because DTMF is passed inband to asterisk by the mISDN channel
and passed out of band once again after the release of the DTMF
tone. Passing it inband is wrong. Neither an analog channel nor
SIP channel passes DTMF inband if configured to inband DTMF.
Analog and SIP channels filter out the DTMF tones because they
use the voice frames returned by ast_dsp_process. But chan_misdn
passes the unfiltered input voice frames instead. To overcome one
aspect of the problem, the doubling of DTMF tones when two mISDN
channels are directly bridged, someone made an 'optimization',
where in that case the DTMF tone passed out-of-band to the peer
channel is not translated to an inband tone at the transmit side.
This optimization is bad because it does not work in general. For
example, analog channels or mISDN channels when bridged through
an intermediary local channel will generate DTMF tones from
out-of-band information. Also, of course, it must not be done
when there is no inband DTMF available. This patch fixes the
issue. Now chan_misdn will filter the received inband DTMF signal
the same as other channel types. Another change included: No need
to build an extra translation path because ast_process_dsp does
it if required. Patches: misdn-dtmf.patch JIRA ABE-2080
................ ................
2010-02-19 17:41 +0000 [r247916] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 247915 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r247915 |
dvossel | 2010-02-19 11:40:26 -0600 (Fri, 19 Feb 2010) | 7 lines
handle_request_invite revise comment, fix coding guideline issues
I'm working with this code right now trying to analyze a
deadlock. This change is just to clean up a few things before I
make a more complex patch. ........
2010-02-18 23:15 +0000 [r247792-247845] Tilghman Lesher <tlesher@digium.com>
* res/res_speech.c, /: Merged revisions 247841 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r247841 |
tilghman | 2010-02-18 17:13:46 -0600 (Thu, 18 Feb 2010) | 7 lines
Revert an errant part of a previous cleanup, to fix a memory
corruption issue. (closes issue #16368) Reported by: thirionjwf
Patches: res_speech.c.patch uploaded by thirionjwf (license 955)
........
* /, channels/chan_sip.c: Merged revisions 247787 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r247787 |
tilghman | 2010-02-18 15:42:53 -0600 (Thu, 18 Feb 2010) | 17
lines If the peer record is from realtime, it could be set to 0,
due to MySQL not representing NULL well in integer columns. NULL
means the value is not specified for the column, which normally
means the driver uses whatever is the default value. However, on
MySQL, placing a NULL in either a float or integer column results
in a retrieval of the 0 value. Hence, users get an errant error
on load. This patch suppresses that error and makes the value as
if it was not there. Note that this cannot be done in the
realtime driver, because the lack of difference between NULL and
0 can only be intepreted correctly by the driver itself. If we
did it in the realtime driver, then it would be effectively
impossible to set any realtime field to 0, because it would act
as if the field were unspecified and possibly take on a different
value. (closes issue #16683) Reported by: wdoekes ........
2010-02-18 21:25 +0000 [r247737-247776] David Vossel <dvossel@digium.com>
* /, bridges/bridge_softmix.c: Merged revisions 247770 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r247770 | dvossel | 2010-02-18 15:23:48 -0600 (Thu, 18 Feb 2010)
| 9 lines fixes confbridge crash when no timing module is loaded.
(closes issue #16471) Reported by: kjotte Patches: M16471.diff
uploaded by junky (license 177) Tested by: kjotte, junky ........
* apps/app_queue.c, /: Merged revisions 247736 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r247736 |
dvossel | 2010-02-18 14:58:41 -0600 (Thu, 18 Feb 2010) | 7 lines
fixes Queue with C option crash (closes issue #16475) Reported
by: okrief Patches: queue_crash.diff uploaded by dvossel (license
671) ........
2010-02-18 19:41 +0000 [r247653] Matthew Nicholson <mnicholson@digium.com>
* /, main/features.c: Merged revisions 247652 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r247652 | mnicholson | 2010-02-18 13:39:37 -0600 (Thu, 18 Feb
2010) | 13 lines Merged revisions 247651 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r247651 | mnicholson | 2010-02-18 13:38:09 -0600 (Thu, 18 Feb
2010) | 6 lines Copy the calling party's account code to the
called party if they don't already have one. (closes issue
#16331) Reported by: bluefox Tested by: mnicholson ........
................
2010-02-18 16:58 +0000 [r247506-247512] Leif Madsen <lmadsen@digium.com>
* README-SERIOUSLY.bestpractices.txt: Merged revisions 247509 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r247509 | lmadsen | 2010-02-18 11:54:43 -0500
(Thu, 18 Feb 2010) | 9 lines Merged revisions 247508 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r247508 | lmadsen | 2010-02-18 11:53:44 -0500 (Thu, 18
Feb 2010) | 1 line Add additional link to best practices document
per jsmith. ........ ................
* README-SERIOUSLY.bestpractices.txt (added): Merged revisions
247503 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r247503 | lmadsen | 2010-02-18 11:41:04 -0500 (Thu, 18 Feb 2010)
| 18 lines Merged revisions 247502 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r247502 | lmadsen | 2010-02-18 11:38:17 -0500 (Thu, 18 Feb 2010)
| 10 lines Add best practices documentation. (issue #16808)
Reported by: lmadsen (issue #16810) Reported by: Nick_Lewis
Tested by: lmadsen Review:
https://reviewboard.asterisk.org/r/507/ ........ ................
2010-02-18 04:21 +0000 [r247426] Russell Bryant <russell@digium.com>
* sounds/Makefile, Makefile, /: Merged revisions 247423 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r247423 | russell | 2010-02-17 22:20:11 -0600
(Wed, 17 Feb 2010) | 17 lines Merged revisions 247422 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r247422 | russell | 2010-02-17 22:19:01 -0600 (Wed, 17 Feb 2010)
| 10 lines Tweak argument handling for wget in the sounds
Makefile. 1) Fix the check to see if we are using wget to not be
full of fail. The configure script populates this variable with
the absolute path to wget if it is found, so it didn't work. 2)
Allow some extra arguments to be passed in for wget. This is just
a simple change to allow our Bamboo build script to tell wget to
be quiet and not fill up our logs with download status output.
........ ................
2010-02-17 21:32 +0000 [r246989-247337] Mark Michelson <mmichelson@digium.com>
* include/asterisk/strings.h, main/strings.c, /: Merged revisions
247335 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r247335 |
mmichelson | 2010-02-17 15:22:40 -0600 (Wed, 17 Feb 2010) | 20
lines Fix two problems in ast_str functions found while writing a
unit test. 1. The documentation for ast_str_set and
ast_str_append state that the max_len parameter may be -1 in
order to limit the size of the ast_str to its current allocated
size. The problem was that the max_len parameter in all cases was
a size_t, which is unsigned. Thus a -1 was interpreted as
UINT_MAX instead of -1. Changing the max_len parameter to be
ssize_t fixed this issue. 2. Once issue 1 was fixed, there was an
off-by-one error in the case where we attempted to write a string
larger than the current allotted size to a string when -1 was
passed as the max_len parameter. When trying to write more than
the allotted size, the ast_str's __AST_STR_USED was set to 1
higher than it should have been. Thanks to Tilghman for quickly
spotting the offending line of code. Oh, and the unit test that I
referenced in the top line of this commit will be added to
reviewboard shortly. Sit tight... ........
* apps/app_queue.c, /: Merged revisions 247169 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r247169 | mmichelson | 2010-02-17 10:24:54 -0600 (Wed, 17 Feb
2010) | 9 lines Merged revisions 247168 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r247168 | mmichelson | 2010-02-17 10:24:17 -0600 (Wed, 17 Feb
2010) | 3 lines Make sure that when autofill is disabled that
callers not in the front of the queue cannot place calls.
........ ................
* main/strings.c, /: Merged revisions 247076 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r247076 |
mmichelson | 2010-02-16 17:44:33 -0600 (Tue, 16 Feb 2010) | 12
lines Add va_end calls to __ast_str_helper. According to the man
page for stdarg(3), "Each invocation of va_copy() must be matched
by a corresponding invocation of va_end() in the same function."
There were several cases in __ast_str_helper where va_copy was
not matched with a corresponding call to va_end. ........
* include/asterisk/strings.h, /: Merged revisions 246985 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r246985 | mmichelson | 2010-02-16 15:15:38 -0600 (Tue,
16 Feb 2010) | 3 lines Add some clarifying documentation to the
ast_str_set and ast_str_append functions. ........
2010-02-16 21:03 +0000 [r246900-246982] David Vossel <dvossel@digium.com>
* main/tcptls.c, /: Merged revisions 246980 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r246980 |
dvossel | 2010-02-16 14:54:48 -0600 (Tue, 16 Feb 2010) | 8 lines
warning message if openssl support is missing while attempting
tls connection (closes issue #16673) Reported by: michaesc
Patches: tls_error_msg.diff uploaded by dvossel (license 671)
........
* main/channel.c: fixes merge error with Monitor calculation fix
* main/channel.c, /: Merged revisions 246899 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r246899 |
dvossel | 2010-02-16 11:07:41 -0600 (Tue, 16 Feb 2010) | 16 lines
fixes sample rate conversion issue with Monitor application When
using ast_seekstream with the read/write streams of a monitor,
the number of samples we are seeking must be of the same rate as
the stream or the jump calculation will be incorrect. This patch
adds logic to correctly convert the number of samples to jump to
the sample rate the read/write stream is using. For example, if
the call is G722 (16khz) and the read/write stream is recording a
8khz wav, seeking 320 samples of 16khz audio is not the same as
seeking 320 samples of 8khz audio when performing the
ast_seekstream on the stream. ABE-2044 ........
2010-02-15 23:45 +0000 [r246713] Tilghman Lesher <tlesher@digium.com>
* Makefile, /: Merged revisions 246710 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r246710 | tilghman | 2010-02-15 17:43:28 -0600 (Mon, 15 Feb 2010)
| 12 lines Merged revisions 246709 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r246709 | tilghman | 2010-02-15 17:42:33 -0600 (Mon, 15 Feb 2010)
| 5 lines Make the menuselect instructions correct by allowing
'make menuselect' to actually solve dependency problems.
(Previously, it would fail out again with the same message about
running 'make menuselect', which was NOT at all helpful.)
........ ................
2010-02-12 23:33 +0000 [r246547] David Vossel <dvossel@digium.com>
* main/channel.c, /: Merged revisions 246546 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r246546 | dvossel | 2010-02-12 17:32:33 -0600 (Fri, 12 Feb 2010)
| 21 lines Merged revisions 246545 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r246545 | dvossel | 2010-02-12 17:30:17 -0600 (Fri, 12 Feb 2010)
| 16 lines lock channel during datastore removal On channel
destruction the channel's datastores are removed and destroyed.
Since there are public API calls to find and remove datastores on
a channel, a lock should be held whenever datastores are removed
and destroyed. This resolves a crash caused by a race condition
in app_chanspy.c. (closes issue #16678) Reported by:
tim_ringenbach Patches: datastore_destroy_race.diff uploaded by
tim ringenbach (license 540) Tested by: dvossel ........
................
2010-02-12 19:08 +0000 [r246464] Jason Parker <jparker@digium.com>
* main/channel.c: Fix some silly formatting that made my head hurt.
2010-02-10 21:28 +0000 [r246199-246207] Tilghman Lesher <tlesher@digium.com>
* /, funcs/func_strings.c: Merged revisions 246204 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r246204 | tilghman | 2010-02-10 15:24:10 -0600 (Wed, 10 Feb 2010)
| 2 lines Fussy compiler on another machine... ........
* /, funcs/func_strings.c: Merged revisions 246200 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r246200 | tilghman | 2010-02-10 15:19:35 -0600 (Wed, 10 Feb 2010)
| 2 lines Fix weird issue with unit tests on optimized build -
turned out to be a signing issue. ........
* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
res/res_agi.c: Merged revisions 246030 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r246030 |
tilghman | 2010-02-10 10:01:28 -0600 (Wed, 10 Feb 2010) | 12
lines Solaris doesn't like outputting a NULL to a %s in format
strings. Detect all platforms that don't like that, either, and
ensure that when documentation is missing, we pass a non-NULL
pointer when outputting the corresponding documentation. (closes
issue #16689) Reported by: bklang Patches:
20100209__issue16689__with_tests.diff.txt uploaded by tilghman
(license 14) Review: https://reviewboard.asterisk.org/r/497/
........
2010-02-10 17:51 +0000 [r246117] David Vossel <dvossel@digium.com>
* apps/app_queue.c, /: Merged revisions 246116 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r246116 | dvossel | 2010-02-10 11:49:34 -0600 (Wed, 10 Feb 2010)
| 14 lines Merged revisions 246115 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r246115 | dvossel | 2010-02-10 11:44:20 -0600 (Wed, 10 Feb 2010)
| 8 lines fixes random deadlock in app_queue with use_weight
during reload (closes issue #16677) Reported by: tim_ringenbach
Patches: app_queue_use_weight_deadlock.diff uploaded by tim
ringenbach (license 540) ........ ................
2010-02-10 16:58 +0000 [r246073] Jeff Peeler <jpeeler@digium.com>
* channels/chan_local.c, /: Merged revisions 246070 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r246070 | jpeeler | 2010-02-10 10:47:37 -0600 (Wed, 10 Feb 2010)
| 22 lines Change channel state on local channels for
busy,answer,ring. Previously local channels channel state never
changed. This became problematic when the state of the other side
of the local channel was lost, for example during a masquerade.
Changing the state of the local channel allows for the scenario
to be detected when the channel state is set to ringing, but the
peer isn't ringing. The specific problem scenario is described in
164201. Although this was noted on one of the issues, here is the
tested dialplan verified to work: exten =>
9700,1,Dial(Local/*9700@default&Local/0009700@default) exten =>
*9700,1,Set(GLOBAL(TESTCHAN)=${CHANNEL:0:${MATH(${LEN(${CHANNEL})}-1):0:2}}1)
exten => *9700,n,wait(3) ;3 works, 1 did not exten =>
*9700,n,Dial(SIP/5001) exten => 0009700,1,Wait(1) ;1 works, 3 did
not exten =>
0009700,n,ChannelRedirect(${TESTCHAN},parkedcalls,701,1) (closes
issue #14992) Reported by: davidw ........
2010-02-10 15:38 +0000 [r245948-246025] Tilghman Lesher <tlesher@digium.com>
* /, funcs/func_strings.c: Merged revisions 246022 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r246022 | tilghman | 2010-02-10 09:36:57 -0600 (Wed, 10 Feb 2010)
| 2 lines Enable warnings on atypical conditions for the FILTER
function (suggested by mmichelson on the -dev list). ........
* configs/extensions.conf.sample, /, funcs/func_strings.c: Merged
revisions 245945 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r245945 | tilghman | 2010-02-10 08:06:12 -0600 (Wed, 10 Feb 2010)
| 9 lines Merged revisions 245944 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r245944 | tilghman | 2010-02-10 07:37:13 -0600 (Wed, 10 Feb 2010)
| 2 lines Include examples of FILTER usage in extension patterns
where a "." may be a risk. ........ ................
2010-02-09 23:11 +0000 [r245794] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c, /: Merged revisions 245793 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r245793 | dvossel | 2010-02-09 17:07:17 -0600
(Tue, 09 Feb 2010) | 18 lines Merged revisions 245792 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r245792 | dvossel | 2010-02-09 16:55:38 -0600 (Tue, 09 Feb 2010)
| 12 lines Fixes iaxs and iaxsl size off by one issue. 2^15 =
32768 which is the maximum allowed iax2 callnumber. Creating the
iaxs and iaxsl array of size 32768 means the maximum callnumber
is actually out of bounds. This causes a nasty crash. (closes
issue #15997) Reported by: exarv Patches: iax_fix.diff uploaded
by dvossel (license 671) ........ ................
2010-02-09 18:09 +0000 [r245732] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_fax.c: Merged revisions 245729 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r245729 |
tilghman | 2010-02-09 12:06:30 -0600 (Tue, 09 Feb 2010) | 8 lines
Ensure frames are only freed once. (closes issue #16361) Reported
by: vlad Patches: 20100208__issue16361.diff.txt uploaded by
tilghman (license 14) Tested by: kenny, bloodoff, misaksen
........
2010-02-09 17:43 +0000 [r245728] Matthew Nicholson <mnicholson@digium.com>
* /, channels/chan_sip.c: Merged revisions 245727 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r245727 |
mnicholson | 2010-02-09 11:40:04 -0600 (Tue, 09 Feb 2010) | 2
lines This commit removes an extra newline in T.38 generated SDP
packets. This bug was caused by the fix introduced in r243860.
(closes issue #16766) Reported by: raivisr Patches:
t38-sdp-newline-fix1.diff uploaded by mnicholson (license 96)
Tested by: raivisr ........
2010-02-09 16:26 +0000 [r245683] Kevin P. Fleming <kpfleming@digium.com>
* /, apps/app_fax.c: Merged revisions 245680 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r245680 |
kpfleming | 2010-02-09 10:24:52 -0600 (Tue, 09 Feb 2010) | 8
lines Don't offer MMR or JBIG transcoding during T.38
negotiation. After further discussion with Steve Underwood, we
should not (yet) be offering to receive MMR or JBIG transcoded
streams from T.38 endpoints. A future spandsp release will
support those features, and then they can be enabled during
negotiation ........
2010-02-08 23:47 +0000 [r245626] Russell Bryant <russell@digium.com>
* /, main/event.c: Merged revisions 245624 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r245624 |
russell | 2010-02-08 17:43:00 -0600 (Mon, 08 Feb 2010) | 5 lines
Fix return value of get_ie_str() and get_ie_str_hash() for
non-existent IE. I found this bug while developing a unit test
for event allocation. Testing is awesome. ........
2010-02-08 22:46 +0000 [r245581] Tilghman Lesher <tlesher@digium.com>
* channels/Makefile, /, main/Makefile: Merged revisions 245578 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r245578 | tilghman | 2010-02-08 16:31:40 -0600 (Mon, 08
Feb 2010) | 12 lines Actually use _ASTLDFLAGS in the main/ and
channels/ Makefiles. They were previously passed correctly, but
they simply weren't used. This caused issues with various
platforms whose builds needed to pass special linker flags via
the configure script. (closes issue #16596) Reported by:
pprindeville Patches: asterisk-1.6-astldflags.patch uploaded by
pprindeville (license 347) Tested by: tilghman ........
2010-02-08 20:43 +0000 [r245500] Jason Parker <jparker@digium.com>
* main/ast_expr2.fl, /, main/ast_expr2f.c: Merged revisions 245497
via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r245497 | qwell | 2010-02-08 14:41:05 -0600
(Mon, 08 Feb 2010) | 11 lines Merged revisions 245496 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r245496 | qwell | 2010-02-08 14:39:50 -0600 (Mon, 08 Feb 2010) |
4 lines Remove reference of documentation in source directory.
People don't always build Asterisk from source (distro packages,
anybody?). ........ ................
2010-02-05 19:27 +0000 [r245097] Jeff Peeler <jpeeler@digium.com>
* contrib/firmware (removed), /, LICENSE: Merged revisions 245090
via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r245090 | jpeeler | 2010-02-05 13:26:22 -0600
(Fri, 05 Feb 2010) | 11 lines Merged revisions 245044 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r245044 | kpfleming | 2010-02-05 12:32:29 -0600 (Fri, 05 Feb
2010) | 5 lines Remove contrib/firmware directory as it is empty
Remove explicit license for IAXy firmware as it is no longer
included in the tree ........ ................
2010-02-05 17:10 +0000 [r244930] Sean Bright <sean@malleable.com>
* main/asterisk.c, /: Merged revisions 244927 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r244927 | seanbright | 2010-02-05 12:05:32 -0500 (Fri, 05 Feb
2010) | 9 lines Merged revisions 244926 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r244926 | seanbright | 2010-02-05 12:03:35 -0500 (Fri, 05 Feb
2010) | 1 line Update main copyright date. ........
................
2010-02-03 19:28 +0000 [r244555] Mark Michelson <mmichelson@digium.com>
* main/sched.c, /: Merged revisions 244547 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r244547 |
mmichelson | 2010-02-03 13:26:53 -0600 (Wed, 03 Feb 2010) | 3
lines Initialize counters in ast_sched_report so that resulting
data is not bogus. ........
2010-02-03 18:47 +0000 [r244508] Tilghman Lesher <tlesher@digium.com>
* channels/chan_dahdi.c, /, main/ast_expr2f.c: Merged revisions
244505 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r244505 |
tilghman | 2010-02-03 12:34:29 -0600 (Wed, 03 Feb 2010) | 8 lines
The chanvar= setting should inherit the entire list of variables,
not just the first one. (closes issue #16359) Reported by: raarts
Patches: dahdi-setvars.diff uploaded by raarts (license 937)
Tested by: raarts ........
2010-02-02 22:29 +0000 [r244445] David Vossel <dvossel@digium.com>
* main/udptl.c, /, channels/chan_sip.c, include/asterisk/udptl.h:
Merged revisions 244443 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r244443 |
dvossel | 2010-02-02 16:27:23 -0600 (Tue, 02 Feb 2010) | 18 lines
fixes crash during T.38 negotiation caused by invalid or missing
FaxMaxDatagram field AST-2010-001 (closes issue #16634) Reported
by: krn (closes issue #16724) Reported by: barthpbx (closes issue
#16517) Reported by: bklang (closes issue #16485) Reported by:
elsto ........
2010-02-02 20:35 +0000 [r244395] Tilghman Lesher <tlesher@digium.com>
* apps/app_dial.c, /: Merged revisions 244393 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r244393 |
tilghman | 2010-02-02 14:32:29 -0600 (Tue, 02 Feb 2010) | 18
lines Properly respect GOSUB_RESULT as to what to do with the
master channel. Previously, we would parse GOSUB_RESULT, but not
actually do anything with it. (closes issue #16686) Reported by:
bklang Patches: app_dial-respect-gosub_result.patch uploaded by
bklang (license 919) (with modifications) ........
2010-02-02 Leif Madsen <lmadsen@digium.com>
* Release Asterisk 1.6.2.2
* AST-2010-001: An attacker attempting to negotiate T.38 over SIP can
remotely crash Asterisk by modifying the FaxMaxDatagram field of
the SDP to contain either a negative or exceptionally large value.
The same crash occurs when the FaxMaxDatagram field is omitted from
the SDP as well.
2010-01-14 Leif Madsen <lmadsen@digium.com>
* Release Asterisk 1.6.2.1
2010-01-08 Leif Madsen <lmadsen@digium.com>
* Release Asterisk 1.6.2.1-rc1
2010-01-07 21:17 +0000 [r238499] Tilghman Lesher <tlesher@digium.com>
* channels/chan_console.c, channels/chan_oss.c, main/poll.c,
channels/chan_usbradio.c, include/asterisk/utils.h, /,
channels/chan_sip.c, channels/chan_alsa.c: Merged revisions
209400 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r209400 |
kpfleming | 2009-07-28 08:49:46 -0500 (Tue, 28 Jul 2009) | 3
lines Define side-effect-safe MIN and MAX macros and remove
duplicate definitions from various files. (closes issue #16251)
Reported by: asgaroth ........
2010-01-07 20:17 +0000 [r238362-238416] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c, /: Merged revisions 238412 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r238412 | dvossel | 2010-01-07 14:15:27 -0600
(Thu, 07 Jan 2010) | 16 lines Merged revisions 238411 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r238411 | dvossel | 2010-01-07 14:14:25 -0600 (Thu, 07 Jan 2010)
| 10 lines fixes crash in "scheduled_destroy" in chan_iax A
signed short was used to represent a callnumber. This is makes it
possible to attempt to access the iaxs array with a negative
index. (closes issue #16565) Reported by: jensvb ........
................
* /, channels/chan_sip.c: Merged revisions 238405 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r238405 |
dvossel | 2010-01-07 14:00:31 -0600 (Thu, 07 Jan 2010) | 8 lines
Change in sip show channels display format allowing more digits
for CID (closes issue #16459) Reported by: Rzadzins Patches:
chan_sip_longer_cid.patch uploaded by Rzadzins (license 953)
........
* apps/app_queue.c, /: Merged revisions 238361 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r238361 |
dvossel | 2010-01-07 12:58:23 -0600 (Thu, 07 Jan 2010) | 8 lines
cli 'queue show' formatting fix. queue name was truncated over 12
characters (closes issue #16078) Reported by: RoadKill Patches:
quequename_limit.patch uploaded by ppyy (license 906) Tested by:
dvossel ........
2010-01-07 09:49 +0000 [r238349] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* configs/sip.conf.sample, /: Merged revisions 238313 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r238313 | tzafrir | 2010-01-07 11:14:57 +0200 (ה', 07 ינו 2010) |
2 lines Document the usefulness of explicit udp:// in the
register string ........
2010-01-06 21:48 +0000 [r238234] Tilghman Lesher <tlesher@digium.com>
* /, funcs/func_cdr.c: Merged revisions 238231 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r238231 | tilghman | 2010-01-06 15:45:17 -0600 (Wed, 06 Jan 2010)
| 11 lines Merged revisions 238230 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r238230 | tilghman | 2010-01-06 15:41:55 -0600 (Wed, 06 Jan 2010)
| 4 lines Revise documentation on disposition values to the
actual values used. (closes issue #16289) Reported by: wdoekes
........ ................
2010-01-06 20:40 +0000 [r238137-238185] Jeff Peeler <jpeeler@digium.com>
* /, apps/app_meetme.c: Merged revisions 238181 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r238181 |
jpeeler | 2010-01-06 14:37:18 -0600 (Wed, 06 Jan 2010) | 8 lines
Fix misreverting from 177158. (closes issue #15725) Reported by:
shanermn Patches: v1-15725.patch uploaded by dimas (license 88)
Tested by: shanermn ........
* /, main/features.c: Merged revisions 238134 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r238134 |
jpeeler | 2010-01-06 13:05:06 -0600 (Wed, 06 Jan 2010) | 10 lines
Fix channel name comparison for bridge application. The channel
name comparison was not comparing the whole string and therefore
if one channel name was a substring of the other, the bridge
would fail. (closes issue #16528) Reported by: telecos82 Patches:
res_features_r236843.diff uploaded by telecos82 (license 687)
........
2010-01-06 15:22 +0000 [r238013] Russell Bryant <russell@digium.com>
* /, apps/app_mp3.c: Merged revisions 238010 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r238010 | russell | 2010-01-06 09:19:10 -0600 (Wed, 06 Jan 2010)
| 14 lines Merged revisions 238009 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r238009 | russell | 2010-01-06 09:18:22 -0600 (Wed, 06 Jan 2010)
| 7 lines Resolve a crash due to an ast_frame not being fully
initialized. (closes issue #16531) Reported by: john8675309
(closes SWP-615) ........ ................
2010-01-06 06:54 +0000 [r237969] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_sip.c: Merged revisions 237968 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r237968 |
tilghman | 2010-01-06 00:53:23 -0600 (Wed, 06 Jan 2010) | 2 lines
Whoa, duplicate setting (dead code). ........
2010-01-05 23:10 +0000 [r237924] Kinsey Moore <kmoore@digium.com>
* apps/app_test.c: Add a wait to ensure TestServer thinks it has
finished sending the final digit. This was previously committed
to 1.4, 1.6.0, 1.6.1, and trunk just after 1.6.2 was created (and
missed). 1.6.2 also needs this patch to resolve the bug. (closes
issue #16550) Reported by: opticron Patches: apptest.diff
uploaded by opticron (license 267)
2010-01-05 23:09 +0000 [r237840-237921] David Vossel <dvossel@digium.com>
* apps/app_queue.c, /: Merged revisions 237920 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r237920 |
dvossel | 2010-01-05 17:08:50 -0600 (Tue, 05 Jan 2010) | 16 lines
fixes holdtime playback issue in app_queue When reporting hold
time, the number of seconds should be mod 60. Otherwise audio
playback could be something like "2 minutes 123 seconds" rather
than "2 minutes 3 seconds". Also, the "minute" sound file is
missing, so for the moment until that file can be created the
"minutes" file is used instead. (closes issue #16168) Reported
by: nickilo Patches: patch-unified-trunk-rev-222176 uploaded by
nickilo (license ) Tested by: nickilo, wonderg ........
* main/pbx.c, /: Merged revisions 237839 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r237839 |
dvossel | 2010-01-05 13:29:47 -0600 (Tue, 05 Jan 2010) | 19 lines
fixes subscriptions being lost after 'module reload' During a
module reload if multiple extension configs are present, such as
both extensions.conf and extensions.ael, watchers for one
config's hints will be lost during the merging of the other
config. This happens because hint watchers are only preserved for
the current config being merged. The old context list is
destroyed after the merging takes place, meaning any watchers
that were not perserved will be removed. Now all hints are
preserved during merging regardless of what config file is being
merged. These hints are only restored if they are present within
the new context list. (closes issue #16093) Reported by: jlaroff
........
2010-01-05 17:25 +0000 [r237743] Russell Bryant <russell@digium.com>
* /, main/utils.c: Merged revisions 237699 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r237699 | russell | 2010-01-05 11:16:01 -0600 (Tue, 05 Jan 2010)
| 14 lines Merged revisions 237697 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r237697 | russell | 2010-01-05 11:13:28 -0600 (Tue, 05 Jan 2010)
| 7 lines Change a NOTICE log message to DEBUG where it belongs.
(closes issue #16479) Reported by: alexrecarey (closes SWP-577)
........ ................
2010-01-05 16:09 +0000 [r237657] Michiel van Baak <michiel@vanbaak.info>
* apps/app_mixmonitor.c, /: Merged revisions 237656 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r237656 | mvanbaak | 2010-01-05 17:08:12 +0100 (Tue, 05 Jan 2010)
| 6 lines Make CLI command 'mixmonitor start|stop <channel> work
again. (closes issue #16534) Reported by: jlaguilar Fix as
suggested by jlaguilar in the bugreport ........
2010-01-04 21:52 +0000 [r237409-237577] Tilghman Lesher <tlesher@digium.com>
* /, main/say.c: Merged revisions 237574 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r237574 | tilghman | 2010-01-04 15:48:20 -0600 (Mon, 04 Jan 2010)
| 13 lines Merged revisions 237573 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r237573 | tilghman | 2010-01-04 15:45:46 -0600 (Mon, 04 Jan 2010)
| 6 lines Bounds checking for input string (closes issue #16407)
Reported by: qwell Patches: 20100104__issue16407.diff.txt
uploaded by tilghman (license 14) ........ ................
* main/pbx.c, /: Merged revisions 237494 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r237494 | tilghman | 2010-01-04 14:59:01 -0600 (Mon, 04 Jan 2010)
| 15 lines Merged revisions 237493 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r237493 | tilghman | 2010-01-04 14:57:35 -0600 (Mon, 04 Jan 2010)
| 8 lines Regression in issue #15421 - Pattern matching (closes
issue #16482) Reported by: wdoekes Patches:
astsvn-16482-betterfix.diff uploaded by wdoekes (license 717)
20091223__issue16482.diff.txt uploaded by tilghman (license 14)
Tested by: wdoekes, tilghman ........ ................
* main/config.c, /: Merged revisions 237414 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r237414 |
tilghman | 2010-01-04 13:03:20 -0600 (Mon, 04 Jan 2010) | 2 lines
Oops, didn't compile (thanks, kpfleming) ........
* main/config.c, /: Merged revisions 237410 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r237410 |
tilghman | 2010-01-04 12:42:10 -0600 (Mon, 04 Jan 2010) | 7 lines
Further reduce the encoded blank values back to blank in the
realtime API. (closes issue #16533) Reported by: sergee Patches:
200100104__issue16533.diff.txt uploaded by tilghman (license 14)
Tested by: sergee ........
* main/pbx.c, /, res/res_agi.c, include/asterisk/channel.h: Merged
revisions 237406 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r237406 | tilghman | 2010-01-04 12:28:28 -0600 (Mon, 04 Jan 2010)
| 23 lines Merged revisions 237405 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r237405 | tilghman | 2010-01-04 12:19:00 -0600 (Mon, 04 Jan 2010)
| 16 lines Add a flag to disable the Background behavior, for AGI
users. This is in a section of code that relates to two other
issues, namely issue #14011 and issue #14940), one of which was
the behavior of Background when called with a context argument
that matched the current context. This fix broke FreePBX,
however, in a post-Dial situation. Needless to say, this is an
extremely difficult collision of several different issues. While
the use of an exception flag is ugly, fixing all of the issues
linked is rather difficult (although if someone would like to
propose a better solution, we're happy to entertain that
suggestion). (closes issue #16434) Reported by: rickead2000
Patches: 20091217__issue16434.diff.txt uploaded by tilghman
(license 14) 20091222__issue16434__1.6.1.diff.txt uploaded by
tilghman (license 14) Tested by: rickead2000 ........
................
2010-01-04 16:50 +0000 [r237328] David Vossel <dvossel@digium.com>
* apps/app_queue.c, /: Merged revisions 237327 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r237327 |
dvossel | 2010-01-04 10:39:11 -0600 (Mon, 04 Jan 2010) | 10 lines
app_queue segfaults if realtime field uniqueid is NULL (closes
issue #16385) Reported by: haakon Patches: app_queue.c.patch
uploaded by haakon (license 880) app_queue.c.patch_v2 uploaded by
dvossel (license 671) Tested by: haakon ........
2010-01-04 16:27 +0000 [r237326] Jeff Peeler <jpeeler@digium.com>
* /, res/res_agi.c: Merged revisions 237323 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r237323 |
jpeeler | 2010-01-04 10:24:51 -0600 (Mon, 04 Jan 2010) | 5 lines
Fix timeout for AGI command speech recognize. (closes issue
#16297) Reported by: semond ........
2010-01-04 16:21 +0000 [r237322] Tilghman Lesher <tlesher@digium.com>
* channels/chan_local.c, /: Merged revisions 237319 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r237319 | tilghman | 2010-01-04 10:20:03 -0600
(Mon, 04 Jan 2010) | 10 lines Merged revisions 237318 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r237318 | tilghman | 2010-01-04 10:18:59 -0600 (Mon, 04 Jan 2010)
| 3 lines It's also possible for the Local channel to directly
execute an Application. Reviewboard:
https://reviewboard.asterisk.org/r/452/ ........ ................
2010-01-02 10:03 +0000 [r237139] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Merged revisions 237136 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r237136 | oej | 2010-01-02 10:54:22 +0100 (Lör, 02 Jan 2010) | 10
lines Merged revisions 237135 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r237135 | oej | 2010-01-02 10:52:30 +0100 (Lör, 02 Jan 2010) | 2
lines Release memory of the contact acl before unloading module
........ ................
2009-12-30 22:00 +0000 [r236985] Tilghman Lesher <tlesher@digium.com>
* channels/chan_local.c, /: Merged revisions 236982 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r236982 | tilghman | 2009-12-30 15:59:18 -0600
(Wed, 30 Dec 2009) | 16 lines Merged revisions 236981 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r236981 | tilghman | 2009-12-30 15:57:10 -0600 (Wed, 30 Dec 2009)
| 9 lines Don't queue frames to channels that have no means to
process them. (closes issue #15609) Reported by: aragon Patches:
20091230__issue16521__1.4__chan_local_only.diff.txt uploaded by
tilghman (license 14) Tested by: aragon Review:
https://reviewboard.asterisk.org/r/452/ ........ ................
2009-12-30 21:13 +0000 [r236899-236905] Jeff Peeler <jpeeler@digium.com>
* /, utils/ael_main.c: Merged revisions 236902 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r236902 |
jpeeler | 2009-12-30 15:09:28 -0600 (Wed, 30 Dec 2009) | 2 lines
One more LOW_MEMORY compile fix. ........
* main/cli.c, /: Merged revisions 236893 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r236893 |
jpeeler | 2009-12-30 14:34:41 -0600 (Wed, 30 Dec 2009) | 11 lines
Fix compiling with LOW_MEMORY. Modified handle_verbose to be
LOW_MEMORY aware. (closes issue #16381) Reported by:
michael_iedema Patches: ast_complete_source_filename.patch
uploaded by michael iedema (license 942) modified by me ........
2009-12-30 17:56 +0000 [r236804-236850] Tilghman Lesher <tlesher@digium.com>
* /, cdr/cdr_adaptive_odbc.c: Merged revisions 236847 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r236847 | tilghman | 2009-12-30 11:53:29 -0600 (Wed, 30 Dec 2009)
| 4 lines When the field is blank, don't warn about the field
being unable to be coerced, just skip the column. (closes
http://lists.digium.com/pipermail/asterisk-dev/2009-December/041362.html)
Reported by Nic Colledge on the -dev list, fixed by me. ........
* /, channels/chan_sip.c: Merged revisions 236802 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r236802 |
tilghman | 2009-12-29 17:05:45 -0600 (Tue, 29 Dec 2009) | 7 lines
Shut down the SIP session timers more gracefully, in order to
prevent a possible crash. (closes issue #16452) Reported by:
corruptor Patches: 20091221__issue16452.diff.txt uploaded by
tilghman (license 14) Tested by: corruptor ........
2009-12-28 22:13 +0000 [r236716] Jason Parker <jparker@digium.com>
* main/ast_expr2.c, /, main/ast_expr2.y: Merged revisions 236713
via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r236713 | qwell | 2009-12-28 16:09:40 -0600 (Mon, 28 Dec
2009) | 8 lines Allow "REMAINDER" to function properly in
expressions. (closes issue #16427) Reported by: wdoekes Patches:
ast16-reminder-remainder.patch uploaded by wdoekes (license 717)
Tested by: wdoekes ........
2009-12-28 17:40 +0000 [r236670] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c, /: Merged revisions 236667 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r236667 | tilghman | 2009-12-28 11:37:46 -0600 (Mon, 28 Dec 2009)
| 4 lines Use recommended option, not deprecated option. (closes
issue #16515) Reported by: ManChicken ........
2009-12-28 15:31 +0000 [r236513-236635] Sean Bright <sean@malleable.com>
* include/asterisk/threadstorage.h, /, configure,
include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
236613 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r236613 | seanbright | 2009-12-28 10:22:54 -0500 (Mon, 28 Dec
2009) | 14 lines Merged revisions 236585 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r236585 | seanbright | 2009-12-28 10:12:08 -0500 (Mon, 28 Dec
2009) | 7 lines Try a test compile to see if PTHREAD_ONCE_INIT
requires extra braces. There was conditional code (based on build
platform) to optioinally wrap PTHREAD_ONCE_INIT in braces that
was removed since it is fixed in newer versions of
Solaris/OpenSolaris, but I am still running into it on Solaris 10
x86 so add a configure-time check for it. ........
................
* /, apps/app_meetme.c: Merged revisions 236510 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r236510 | seanbright | 2009-12-28 07:44:58 -0500 (Mon, 28 Dec
2009) | 19 lines Merged revisions 236509 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r236509 | seanbright | 2009-12-28 07:43:36 -0500 (Mon, 28 Dec
2009) | 12 lines Avoid a crash with large numbers of MeetMe
conferences. Similar to changes made to Queue(), when we have
large numbers of conferences in meetme.conf (1000s) and we use
alloca()/strdupa(), we can blow out the stack and crash, so
instead just use a single fixed buffer. (closes issue #16509)
Reported by: Kashif Raza Patches: 20091223_16509.patch uploaded
by seanbright (license 71) Tested by: seanbright ........
................
2009-12-27 18:22 +0000 [r236437] Tilghman Lesher <tlesher@digium.com>
* contrib/init.d/rc.debian.asterisk, /: Merged revisions 236434 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r236434 | tilghman | 2009-12-27 12:20:53 -0600
(Sun, 27 Dec 2009) | 9 lines Merged revisions 236433 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r236433 | tilghman | 2009-12-27 12:19:38 -0600 (Sun, 27
Dec 2009) | 2 lines Turn on colors in the daemon, since there's
many requests for it on Ubuntu. ........ ................
2009-12-26 15:32 +0000 [r236361] Kevin P. Fleming <kpfleming@digium.com>
* sounds/Makefile, /: Merged revisions 236358 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r236358 | kpfleming | 2009-12-26 09:27:44 -0600 (Sat, 26 Dec
2009) | 9 lines Merged revisions 236357 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r236357 | kpfleming | 2009-12-26 09:26:17 -0600 (Sat, 26 Dec
2009) | 1 line update to latest releases with zero uid/gid
........ ................
2009-12-23 18:27 +0000 [r236189-236303] Tilghman Lesher <tlesher@digium.com>
* apps/app_stack.c, /: Merged revisions 236300 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r236300 |
tilghman | 2009-12-23 12:25:27 -0600 (Wed, 23 Dec 2009) | 7 lines
AGI may be invoked from outside the dialplan (closes issue
#16510) Reported by: atis Patches: 20091223__issue16510.diff.txt
uploaded by tilghman (license 14) Tested by: atis ........
* /, res/res_agi.c: Merged revisions 236186 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r236186 | tilghman | 2009-12-22 21:07:48 -0600 (Tue, 22 Dec 2009)
| 11 lines Merged revisions 236184 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r236184 | tilghman | 2009-12-22 20:55:24 -0600 (Tue, 22 Dec 2009)
| 4 lines If EXEC only gets a single argument, don't crash when
the second is used. (closes issue #16504) Reported by: bklang
........ ................
2009-12-22 17:04 +0000 [r236064] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 236063 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r236063 | dvossel | 2009-12-22 11:00:08 -0600 (Tue, 22 Dec 2009)
| 18 lines Merged revisions 236062 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r236062 | dvossel | 2009-12-22 10:58:19 -0600 (Tue, 22 Dec 2009)
| 11 lines fixes issue with p->method incorrectly set to ACK It
is possible for a second ACK to come in for a retransmitted
message. If an ack does not match an unacked message in our
queue, restore the previous p->method as this ACK is completely
ignored. (closes issue #16295) Reported by: omolenkamp Patches:
issue16295_v2.diff uploaded by dvossel (license 671) ........
................
2009-12-21 19:58 +0000 [r235944] Jeff Peeler <jpeeler@digium.com>
* res/res_monitor.c, /: Merged revisions 235941 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r235941 | jpeeler | 2009-12-21 13:54:20 -0600 (Mon, 21 Dec 2009)
| 20 lines Merged revisions 235940 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r235940 | jpeeler | 2009-12-21 13:43:41 -0600 (Mon, 21 Dec 2009)
| 13 lines Change Monitor to not assume file to write to does not
contain pathing. 227944 changed the fname_base argument to always
append the configured monitor path. This change was necessary to
properly compare files for uniqueness. If a full path is given
though, nothing needs to be appended and that is handled
correctly now. (closes issue #16377) (closes issue #16376)
Reported by: bcnit Patches: res_monitor.c-issue16376-1.patch
uploaded by dant (license 670) ........ ................
2009-12-21 17:11 +0000 [r235826] Tilghman Lesher <tlesher@digium.com>
* /, main/features.c: Merged revisions 235822 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r235822 | tilghman | 2009-12-21 11:00:46 -0600 (Mon, 21 Dec 2009)
| 15 lines Merged revisions 235821 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r235821 | tilghman | 2009-12-21 10:45:03 -0600 (Mon, 21 Dec 2009)
| 8 lines Send parking lot announcement to the channel which
parked the call, not the park-ee. (closes issue #16234) Reported
by: yeshuawatso Patches: 20091210__issue16234.diff.txt uploaded
by tilghman (license 14) 20091221__issue16234__1.4.diff.txt
uploaded by tilghman (license 14) Tested by: yeshuawatso ........
................
2009-12-20 08:58 +0000 [r235775] Alec L Davis <sivad.a@paradise.net.nz>
* main/dsp.c: restarts busydetector (if enabled) when DTMF is
received after call is bridged. (closes issue #16389) Reported
by: alecdavis Tested by: alecdavis Patch
dtmf_busydetector.diff2.txt uploaded by alecdavis (license 585)
2009-12-18 23:04 +0000 [r235665] Jeff Peeler <jpeeler@digium.com>
* main/channel.c, /, include/asterisk/cdr.h: Merged revisions
235660 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r235660 | jpeeler | 2009-12-18 16:51:37 -0600 (Fri, 18 Dec 2009)
| 55 lines Merged revisions 235635 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r235635 | jpeeler | 2009-12-18 16:29:51 -0600 (Fri, 18 Dec 2009)
| 48 lines Correct CDR dispositions for BUSY/FAILED This patch is
simple in that it reorders the disposition defines so that the
fix for issue 12946 works properly (the default CDR disposition
was changed to AST_CDR_NOANSWER). Also, the
AST_CDR_FLAG_ORIGINATED flag was set in ast_call to ensure all
CDR records are written. The side effects of CDR changes are
scary, so I'm documenting the test cases performed to attempt to
catch any regressions. The following tests were all performed
using 1.4 rev 195881 vs head (235571) + patch: A calls B C calls
B (busy) Hangup C Hangup A (Both SIP and features) A calls B A
blind transfers to C Hangup C (Both SIP and features) A calls B A
attended transfers to C Hangup C A calls B A attended transfers
to C (SIP) C blind transfers to A (features) Hangup A All of the
test scenario CDRs matched. The following tests were performed
just with the patch to ensure proper operation (with
unanswered=yes): exten =>s,1,Answer exten =>s,n,ResetCDR(w) exten
=>s,n,ResetCDR(w) exten =>s,1,ResetCDR(w) exten =>s,n,ResetCDR(w)
(closes issue #16180) Reported by: aatef Patches: bug16180.patch
uploaded by jpeeler (license 325) ........ ................
2009-12-18 22:42 +0000 [r235576-235659] Tilghman Lesher <tlesher@digium.com>
* /, configure, configure.ac: Merged revisions 235656 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r235656 | tilghman | 2009-12-18 16:40:46 -0600
(Fri, 18 Dec 2009) | 9 lines Merged revisions 235652 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r235652 | tilghman | 2009-12-18 16:39:30 -0600 (Fri, 18
Dec 2009) | 2 lines Revise verbiage, per #asterisk-dev discussion
........ ................
* /, configure, configure.ac: Merged revisions 235573 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r235573 | tilghman | 2009-12-18 15:19:43 -0600
(Fri, 18 Dec 2009) | 9 lines Merged revisions 235572 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r235572 | tilghman | 2009-12-18 15:18:16 -0600 (Fri, 18
Dec 2009) | 2 lines Point to the typical missing package, not the
cryptic "termcap support". ........ ................
2009-12-17 23:22 +0000 [r235522] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 235521 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r235521 |
file | 2009-12-17 19:21:07 -0400 (Thu, 17 Dec 2009) | 3 lines
Remove some old code for going to the 'fax' extension when a T.38
switchover occurs. This would have already happened when we
detected the CNG tone so this was basically a noop. ........
2009-12-17 Leif Madsen <lmadsen@digium.com>
* Release Asterisk 1.6.2.0
2009-12-09 Leif Madsen <lmadsen@digium.com>
* Release Asterisk 1.6.2.0-rc8
2009-12-08 18:33 +0000 [r233731] Tilghman Lesher <tlesher@digium.com>
* res/res_musiconhold.c, /: Merged revisions 233718 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r233718 | tilghman | 2009-12-08 12:22:44 -0600 (Tue, 08 Dec 2009)
| 8 lines Find another ref leak and change how we manage module
references. (closes issue #16388) Reported by: parisioa Patches:
20091208__issue16388.diff.txt uploaded by tilghman (license 14)
Tested by: parisioa, tilghman Review:
https://reviewboard.asterisk.org/r/442/ ........
2009-12-08 18:04 +0000 [r233694] Russell Bryant <russell@digium.com>
* formats/format_sln16.c, formats/format_wav_gsm.c,
formats/format_siren7.c, formats/format_ilbc.c,
formats/format_vox.c, formats/format_pcm.c,
formats/format_h263.c, formats/format_g723.c,
formats/format_h264.c, formats/format_siren14.c,
formats/format_jpeg.c, formats/format_g726.c,
formats/format_gsm.c, formats/format_g729.c, /,
formats/format_sln.c, formats/format_wav.c,
formats/format_ogg_vorbis.c: Merged revisions 233692 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r233692 | russell | 2009-12-08 12:00:16 -0600 (Tue, 08 Dec 2009)
| 16 lines Set a module load priority for format modules. A
recent change to app_voicemail made it such that the module now
assumes that all format modules are available while processing
voicemail configuration. However, when autoloading modules, it
was possible that app_voicemail was loaded before the format
modules. Since format modules don't depend on anything, set a
module load priority on them to ensure that they get loaded first
when autoloading. This fix applies to trunk, 1.6.1, and 1.6.2.
The fix for 1.4 and 1.6.0 will require a different approach since
the module load priority functionality is not present in the
module API. (issue #16412) Reported by: jiddings ........
2009-12-08 07:41 +0000 [r233689] TransNexus OSP Development <support@transnexus.com>
* apps/app_osplookup.c: Fixed compile error with OSP Toolkit 3.6.
2009-12-07 23:54 +0000 [r233615] Atis Lezdins <atis@iq-labs.net>
* contrib/valgrind.supp, /: Merged revisions 233577 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r233577 | atis | 2009-12-08 01:10:13 +0200 (Tue, 08 Dec 2009) | 8
lines Fix compatibility with valgrind 3.3 and older. (noticed in
issue #16388) Reported by: parisioa Patches: valgrind.supp
uloaded by atis (license 242) Tested by: atis, parisioa ........
2009-12-07 23:29 +0000 [r233473-233612] David Vossel <dvossel@digium.com>
* /, main/utils.c: Merged revisions 233611 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r233611 |
dvossel | 2009-12-07 17:28:51 -0600 (Mon, 07 Dec 2009) | 4 lines
fixes incorrect logic in ast_uri_encode issue #16299 ........
* /, channels/chan_sip.c: Merged revisions 233472 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r233472 | dvossel | 2009-12-07 12:08:46 -0600 (Mon, 07 Dec 2009)
| 15 lines Merged revisions 233471 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r233471 | dvossel | 2009-12-07 12:07:38 -0600 (Mon, 07 Dec 2009)
| 9 lines fixes missing Contact header angle brackets (closes
issue #16298) Reported by: mgernoth Patches:
reg_parse_issue_1.4.diff uploaded by dvossel (license 671) Tested
by: dvossel ........ ................
2009-12-07 16:16 +0000 [r233396] Matthew Nicholson <mnicholson@digium.com>
* /, channels/chan_sip.c: Merged revisions 233394 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r233394 |
mnicholson | 2009-12-07 10:14:42 -0600 (Mon, 07 Dec 2009) | 8
lines Do not reject SDP packets describing only non audio
streams. (closes issue #16387) Reported by: zalex1953 Patches:
media-level-c-fix1.diff uploaded by mnicholson (license 96)
Tested by: mnicholson, zalex1953 ........
2009-12-04 21:55 +0000 [r233281] David Vossel <dvossel@digium.com>
* configs/iax.conf.sample, /: Merged revisions 233280 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r233280 | dvossel | 2009-12-04 15:54:44 -0600
(Fri, 04 Dec 2009) | 14 lines Merged revisions 233279 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r233279 | dvossel | 2009-12-04 15:54:01 -0600 (Fri, 04 Dec 2009)
| 7 lines clarify requirecalltoken option in iax.sample.conf
(closes issue #16223) Reported by: bklang Patches:
clarify-iax-requirecalltoken.patch uploaded by bklang (license
919) ........ ................
2009-12-04 21:07 +0000 [r233240] Matthias Nick <mnick@digium.com>
* pbx/pbx_config.c, /: Merged revisions 233093 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r233093 |
mnick | 2009-12-04 11:15:47 -0600 (Fri, 04 Dec 2009) | 8 lines
Parse global variables or expressions in hint extensions Parse
global variables or expressions in hint extensions. Like: exten
=> 400,hint,DAHDI/i2/${GLOBAL(var)} (closes issue #16166)
Reported by: rmudgett Tested by: mnick, rmudgett ........
2009-12-04 17:36 +0000 [r233165] David Vossel <dvossel@digium.com>
* apps/app_voicemail.c, /: Merged revisions 233121 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r233121 | dvossel | 2009-12-04 11:22:31 -0600
(Fri, 04 Dec 2009) | 12 lines Merged revisions 233116 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r233116 | dvossel | 2009-12-04 11:21:34 -0600 (Fri, 04 Dec 2009)
| 6 lines document and rename strip_control() in app_voicemail
(closes issue #16291) Reported by: wdoekes ........
................
2009-12-04 17:23 +0000 [r233130] Russell Bryant <russell@digium.com>
* main/channel.c, /: Merged revisions 233100 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r233100 | russell | 2009-12-04 11:18:22 -0600 (Fri, 04 Dec 2009)
| 14 lines Merged revisions 233092 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r233092 | russell | 2009-12-04 11:12:47 -0600 (Fri, 04 Dec 2009)
| 7 lines Only do frame payload check for HOLD frames. This code
was added for helping to debug the source of invalid HOLD frames.
However, a side effect of this is that it will incorrectly report
errors for frames that have an integer payload. Make the check
for this block specific to the HOLD frame case. ........
................
2009-12-04 15:57 +0000 [r233049] Matthias Nick <mnick@digium.com>
* main/dsp.c, /: Merged revisions 233046 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r233046 | mnick | 2009-12-04 09:38:33 -0600 (Fri, 04 Dec 2009) |
17 lines Merged revisions 233014 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r233014 | mnick | 2009-12-04 09:17:03 -0600 (Fri, 04 Dec 2009) |
11 lines Warning message gets displayed only once Added
additional field 'int display_inband_dtmf_warning', which when
set to '1' displays the warning ('Inband DTMF is not supported on
codec %s. Use RFC2833'), and when set to '0' doesn't display the
warning. Otherwise you would get hundreds of warnings every
second. (closes issue #15769) Reported by: falves11 Patches:
patch_15769_14.txt uploaded by mnick (license 874) Tested by:
mnick, falves11 ........ ................
2009-12-03 21:03 +0000 [r232866] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c, /: Merged revisions 232854 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r232854 | tilghman | 2009-12-03 14:47:07 -0600
(Thu, 03 Dec 2009) | 15 lines Merged revisions 232820 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r232820 | tilghman | 2009-12-03 14:10:19 -0600 (Thu, 03 Dec 2009)
| 8 lines Deprecate "cz" in favor of "cs". Also, change the use
of language codes so that language registers as a prefix, rather
than an exact match. (closes issue #16272) Reported by: patrol-cz
Patches: 20091203__issue16272.diff.txt uploaded by tilghman
(license 14) ........ ................
2009-12-03 15:14 +0000 [r232813] David Ruggles <thedavidfactor@gmail.com>
* apps/app_externalivr.c: Merged revisions 232587 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r232587 |
diruggles | 2009-12-02 17:17:22 -0500 (Wed, 02 Dec 2009) | 12
lines Prevent double closing of FDs by EIVR This caused a problem
when asterisk was under heavy load and running both AGI and EIVR
applications. EIVR would close an FD at which point it would be
considered freed and be used by a new AGI instance the second
close would then close the FD now in use by AGI. (closes issue
#16305) Reported by: diLLec Tested by: thedavidfactor, diLLec
Review: https://reviewboard.asterisk.org/r/436/ ........
2009-12-03 00:20 +0000 [r232675-232678] Tilghman Lesher <tlesher@digium.com>
* res/res_musiconhold.c: Oops, really remove it this time
* res/res_musiconhold.c, /: Recorded merge of revisions
232660-232661 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r232660 |
tilghman | 2009-12-02 18:08:55 -0600 (Wed, 02 Dec 2009) | 19
lines Fix multiple issues with musiconhold, which led to classes
not getting destroyed properly. * Classes are now tracked past
removal from the core container, and module removal is actively
prevented until all references are freed. * A hanging reference
stored in the channel has been removed. This could have caused a
mismatch and the music state not properly cleared, if two or more
reloads occurred between MOH being stopped and MOH being
restarted. * In certain circumstances, duplicate classes were
possible. * A race existed at reload time between a process being
killed and the thread responsible for reading from the related
pipe respawning that process. * Several reference counts have
also been corrected. At least one could have caused deleted
classes to stick around forever, consuming resources. This
originally manifested as MOH external processes that were not
killed at reload time. (closes issue #16279, closes issue #16207)
Reported by: parisioa, dcabot Patches:
20091202__issue16279__2.diff.txt uploaded by tilghman (license
14) Tested by: parisioa, tilghman ........ r232661 | tilghman |
2009-12-02 18:09:36 -0600 (Wed, 02 Dec 2009) | 2 lines Remove
debugging line ........
2009-12-02 23:28 +0000 [r232658] David Vossel <dvossel@digium.com>
* CHANGES, /, UPGRADE.txt: Merged revisions 232657 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r232657 | dvossel | 2009-12-02 17:27:45 -0600 (Wed, 02 Dec 2009)
| 6 lines update CHANGES and UPGRADE.txt for early media behavior
change between 1.6.1 and 1.6.2 (closes issue #16212) Reported by:
miki ........
2009-12-02 22:05 +0000 [r232579-232585] Jeff Peeler <jpeeler@digium.com>
* main/manager.c, /: Merged revisions 232582 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r232582 | jpeeler | 2009-12-02 16:02:43 -0600 (Wed, 02 Dec 2009)
| 14 lines Merged revisions 232581 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r232581 | jpeeler | 2009-12-02 15:57:42 -0600 (Wed, 02 Dec 2009)
| 7 lines Send ack (response/message) after receiving manager
action userevent (closes issue #16264) Reported by: dimas
Patches: event-ack.patch uploaded by dimas (license 88) ........
................
* main/manager.c, /: Merged revisions 232576 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r232576 |
jpeeler | 2009-12-02 15:32:50 -0600 (Wed, 02 Dec 2009) | 8 lines
Make manager response to "Action: events" finish with empty line
(closes issue #16275) Reported by: vnovy Patches: manager.c.diff
uploaded by vnovy (license 922) ........
2009-12-02 17:11 +0000 [r232359] Joshua Colp <jcolp@digium.com>
* /, apps/app_amd.c: Merged revisions 232356 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r232356 | file | 2009-12-02 13:06:54 -0400 (Wed, 02 Dec 2009) |
12 lines Merged revisions 232355 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r232355 | file | 2009-12-02 13:04:52 -0400 (Wed, 02 Dec 2009) | 5
lines Fix a bug where if you hung up very quickly after calling
AMD it would overwrite the AMDSTATUS of HANGUP with TOOLONG.
(closes issue #16239) Reported by: CGMChris ........
................
2009-12-02 17:01 +0000 [r232352] David Vossel <dvossel@digium.com>
* /, main/acl.c: Merged revisions 232351 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r232351 | dvossel | 2009-12-02 11:00:15 -0600 (Wed, 02 Dec 2009)
| 12 lines Merged revisions 232350 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r232350 | dvossel | 2009-12-02 10:59:18 -0600 (Wed, 02 Dec 2009)
| 6 lines ast_outaddrfor doesn't do htons() on port, looks odd in
strace. (closes issue #16290) Reported by: wdoekes ........
................
2009-12-02 16:43 +0000 [r232348] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 232345 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r232345 |
file | 2009-12-02 12:40:14 -0400 (Wed, 02 Dec 2009) | 7 lines Add
support for handling the 415 Unsupported media type response like
we do for a 488 Not acceptable here response. (closes issue
#16186) Reported by: atis Patches: sip_t38_response_415.patch
uploaded by atis (license 242) ........
2009-12-02 15:43 +0000 [r232270] David Vossel <dvossel@digium.com>
* funcs/func_groupcount.c, /: Merged revisions 232269 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r232269 | dvossel | 2009-12-02 09:42:54 -0600
(Wed, 02 Dec 2009) | 15 lines Merged revisions 232268 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r232268 | dvossel | 2009-12-02 09:41:36 -0600 (Wed, 02 Dec 2009)
| 9 lines fixes segfault in func_groupcount closes issue #16337)
Reported by: Parantido Patches: issue_16337.diff uploaded by
dvossel (license 671) Tested by: Parantido, dvossel ........
................
2009-12-02 14:55 +0000 [r232232] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 232230 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r232230 |
file | 2009-12-02 10:54:28 -0400 (Wed, 02 Dec 2009) | 5 lines Fix
a bug where a scheduled item ID would get retained on
registrations in a certain scenario causing code to execute
during reload that should not. (issue AST-263) ........
2009-12-02 00:52 +0000 [r232094] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 232091 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r232091 | jpeeler | 2009-12-01 18:45:18 -0600
(Tue, 01 Dec 2009) | 17 lines Merged revisions 232090 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r232090 | jpeeler | 2009-12-01 18:42:58 -0600 (Tue, 01 Dec 2009)
| 10 lines Do not modify the gain settings on data calls. (The
digital flag actually represents a data call.) (closes issue
#15972) Reported by: udosw Patches: transcap_digital_fix.diff.txt
uploaded by alecdavis (license 585) Tested by: alecdavis ........
................
2009-12-01 23:40 +0000 [r232011-232015] Russell Bryant <russell@digium.com>
* /, funcs/func_lock.c: Merged revisions 232012 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r232012 |
russell | 2009-12-01 17:38:34 -0600 (Tue, 01 Dec 2009) | 2 lines
Fix a build error on FreeBSD. ........
* /, main/file.c: Merged revisions 232008 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r232008 | russell | 2009-12-01 17:27:53 -0600 (Tue, 01 Dec 2009)
| 9 lines Merged revisions 232007 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r232007 | russell | 2009-12-01 17:25:36 -0600 (Tue, 01 Dec 2009)
| 2 lines Fix a warning pointed out by buildbot. ........
................
2009-12-01 22:03 +0000 [r231930] Jeff Peeler <jpeeler@digium.com>
* main/channel.c, /: Merged revisions 231927 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r231927 | jpeeler | 2009-12-01 15:54:21 -0600 (Tue, 01 Dec 2009)
| 19 lines Merged revisions 231911 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r231911 | jpeeler | 2009-12-01 15:29:31 -0600 (Tue, 01 Dec 2009)
| 12 lines Fix crash with invalid frame data The crash was
happening as a result of a frame containing an invalid data
pointer, but was set with data length of zero. The few times the
issue was reproduced it _seemed_ that the frame was queued
properly, that is the data pointer was set to NULL. I never could
reproduce the crash so as a last resort the crash has been fixed,
but a check in __ast_read has been added to give as much
information about the source of problematic frames in the future.
(closes issue #16058) Reported by: atis ........ ................
2009-12-01 21:21 +0000 [r231870] David Vossel <dvossel@digium.com>
* main/pbx.c, /: Merged revisions 231867 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r231867 | dvossel | 2009-12-01 15:20:19 -0600 (Tue, 01 Dec 2009)
| 9 lines Merged revisions 231853 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r231853 | dvossel | 2009-12-01 15:14:31 -0600 (Tue, 01 Dec 2009)
| 3 lines WaitExten m option with no parameters generates frame
with zero datalen but non-null data ptr ........ ................
2009-12-01 Leif Madsen <lmadsen@digium.com>
* Release Asterisk 1.6.2.0-rc7
2009-12-01 15:48 +0000 [r231743] Matthew Nicholson <mnicholson@digium.com>
* /, main/file.c: Merged revisions 231741 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r231741 | mnicholson | 2009-12-01 09:47:36 -0600 (Tue, 01 Dec
2009) | 9 lines Merged revisions 231740 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r231740 | mnicholson | 2009-12-01 09:34:57 -0600 (Tue, 01 Dec
2009) | 2 lines Ignore unknown formats in ast_format_str_reduce()
and return an error if no know formats are found. ........
................
2009-11-30 21:59 +0000 [r231695-231696] Kevin P. Fleming <kpfleming@digium.com>
* main/udptl.c, /, channels/chan_sip.c, include/asterisk/udptl.h:
Merged revisions 231692 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r231692 |
kpfleming | 2009-11-30 15:47:42 -0600 (Mon, 30 Nov 2009) | 22
lines Another round of UDPTL stack fixes/improvements: 1) Allow
users of UDPTL stack to associate a character-string tag with a
UDPTL session, so that log/error/debug messages generated by the
UDPTL stack can be 'connected' to the endpoint that caused them
to be generated. 2) Improve comments (and process) of calculating
the far end's maximum IFP size when redundancy mode is in use for
error correction. 3) When an IFP larger than the calculated 'far
max IFP' size is presented for writing, truncate it rather than
putting in the buffer and allowing the buffer to overflow; this
will cause the ends to retrain to a lower bit rate that produces
IFPs of an appropriate size if possible, and if not possible, the
FAX transfer will fail completely. In these cases, it is due to
the one endpoint supplying a T38FaxMaxDatagram value that is
improperly calculated and is too low to be of use; we have
configuration options available to override this behavior. 4)
Eliminate use of T38FaxMaxDatagram value in udptl.conf; it is no
longer needed. ........
* pbx/pbx_config.c: Backport a tiny fix from trunk that makes GCC
4.4.x happier.
2009-11-30 21:36 +0000 [r231689] Matthew Nicholson <mnicholson@digium.com>
* apps/app_voicemail.c, include/asterisk/file.h, /, main/file.c,
main/app.c: Merged revisions 231688 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r231688 | mnicholson | 2009-11-30 15:31:55 -0600 (Mon, 30 Nov
2009) | 15 lines Merged revisions 231614 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r231614 | mnicholson | 2009-11-30 15:11:44 -0600 (Mon, 30 Nov
2009) | 8 lines Remove duplicate entries from voicemail format
lists. This prevents app_voicemail from entering an infinite loop
when the same format is specified twice in the format list.
(closes issue #15625) Reported by: Shagg63 Tested by: mnicholson
Review: https://reviewboard.asterisk.org/r/429/ ........
................
2009-11-30 20:47 +0000 [r231605] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 231602 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r231602 |
file | 2009-11-30 16:44:30 -0400 (Mon, 30 Nov 2009) | 5 lines
When receiving SDP that matches the version of the last one do
not treat it as a fatal error. (closes issue #16238) Reported by:
seandarcy ........
2009-11-30 18:57 +0000 [r231505-231558] David Vossel <dvossel@digium.com>
* apps/app_queue.c, /: Merged revisions 231556 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r231556 |
dvossel | 2009-11-30 12:55:07 -0600 (Mon, 30 Nov 2009) | 11 lines
app_queue crashes randomly, often during call-transfers This
patch adds a ref to the queue_ent object's parent call_queue in
queue_exec() so the call_queue won't be destroyed while the the
queue_ent still holds a pointer to it. (closes issue 0015686)
Tested by: dvossel, aragon ........
* main/rtp.c, /: Merged revisions 231491 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r231491 | dvossel | 2009-11-30 11:28:28 -0600 (Mon, 30 Nov 2009)
| 17 lines Merged revisions 231441 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r231441 | dvossel | 2009-11-30 11:14:08 -0600 (Mon, 30 Nov 2009)
| 11 lines fixes crash caused by RTP comfort noise payload
greater than 24 bytes AST-2009-010 (closes issue #16242) Reported
by: amorsen Patches: issue16242.diff uploaded by oej (license
306) Tested by: amorsen, oej, dvossel ........ ................
2009-11-25 22:34 +0000 [r231302] Tilghman Lesher <tlesher@digium.com>
* main/channel.c, /: Merged revisions 231299 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r231299 | tilghman | 2009-11-25 16:33:02 -0600 (Wed, 25 Nov 2009)
| 9 lines Merged revisions 231298 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r231298 | tilghman | 2009-11-25 16:31:57 -0600 (Wed, 25 Nov 2009)
| 2 lines After a frame duplication failure, unlock the channel
before returning. ........ ................
2009-11-25 15:48 +0000 [r231191] Matthew Nicholson <mnicholson@digium.com>
* /, pbx/pbx_lua.c: Merged revisions 231189 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r231189 |
mnicholson | 2009-11-25 09:42:48 -0600 (Wed, 25 Nov 2009) | 4
lines Load pbx_lua with global symbols to allow linking with
other lua libraries. Found by Maxim Litnitskiy. ........
2009-11-24 20:36 +0000 [r231136] Tilghman Lesher <tlesher@digium.com>
* apps/app_queue.c, /: Merged revisions 231134 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r231134 |
tilghman | 2009-11-24 14:31:28 -0600 (Tue, 24 Nov 2009) | 7 lines
Found a few places where queue refcounts were counted
incorrectly. Also add debug statements. (closes issue #15982,
closes issue #15984) Reported by: atis Patches:
20091111__issue15982.diff.txt uploaded by tilghman (license 14)
Tested by: atis ........
2009-11-24 18:54 +0000 [r231098] Jeff Peeler <jpeeler@digium.com>
* /, main/features.c: Merged revisions 231095 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r231095 |
jpeeler | 2009-11-24 12:50:36 -0600 (Tue, 24 Nov 2009) | 11 lines
Fix erroneous hangup extension execution ast_spawn_extension
behaves differently from 1.4 in that hangups and extensions that
do not exist do not return an error, whereas in 1.6 it does. This
is now taken into account so that the AST_FLAG_BRIDGE_HANGUP_RUN
flag gets set properly. (closes issue #16106) Reported by:
ajohnson Tested by: ajohnson ........
2009-11-23 15:48 +0000 [r230884] Joshua Colp <jcolp@digium.com>
* configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions
230881 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r230881 |
file | 2009-11-23 09:45:45 -0600 (Mon, 23 Nov 2009) | 7 lines
Change fax detection in chan_sip so it behaves as one would
expect. Internally the way T.38 is negotiated has changed and the
option no longer reflects a behavior that is valid. It will now
look for a CNG tone on received calls and if present send the
call to the 'fax' extension. It is then up to the application or
channel to request the switch over to T.38. ........
2009-11-23 15:38 +0000 [r230796-230880] Kevin P. Fleming <kpfleming@digium.com>
* /, channels/chan_sip.c: Merged revisions 230877 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r230877 | kpfleming | 2009-11-23 09:34:16 -0600 (Mon, 23 Nov
2009) | 9 lines Merged revisions 230839 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r230839 | kpfleming | 2009-11-23 09:09:24 -0600 (Mon, 23 Nov
2009) | 1 line Correct fix for issue #16268... the reporter's
original patch was very close to correct. ........
................
* /, channels/chan_sip.c: Merged revisions 230773 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r230773 | kpfleming | 2009-11-23 08:15:48 -0600 (Mon, 23 Nov
2009) | 12 lines Merged revisions 230772 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r230772 | kpfleming | 2009-11-23 08:13:56 -0600 (Mon, 23 Nov
2009) | 5 lines Ensure that SDP parsing does not ignore the last
line of the SDP. (closes issue #16268) Reported by: sgimeno
........ ................
2009-11-20 22:36 +0000 [r230727] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c, /: Merged revisions 230726 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r230726 | dvossel | 2009-11-20 16:35:54 -0600 (Fri, 20 Nov 2009)
| 7 lines fixes iax2 show cache locking error, thanks alecdavis!
(closes issue #16094) Reported by: alecdavis Patches:
bug16094.diff.txt uploaded by alecdavis (license 585) Tested by:
alecdavis, dvossel ........
2009-11-20 21:07 +0000 [r230629] Matthew Nicholson <mnicholson@digium.com>
* /, main/features.c: Merged revisions 230628 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r230628 | mnicholson | 2009-11-20 15:01:10 -0600 (Fri, 20 Nov
2009) | 15 lines Merged revisions 230627 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r230627 | mnicholson | 2009-11-20 14:53:06 -0600 (Fri, 20 Nov
2009) | 8 lines Copy the peer CDR's userfield to the bridge CDR
if it exists. This is necessary for the recordagentcalls option
in chan_agent to store the recorded file name in the bridge CDR.
(closes issue #14590) Reported by: msetim Patches:
queue_agent_userfield.patch uploaded by Laureano (license 265)
Tested by: Laureano, mnicholson ........ ................
2009-11-20 17:31 +0000 [r230510-230585] David Vossel <dvossel@digium.com>
* main/audiohook.c, /, include/asterisk/audiohook.h: Merged
revisions 230583 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r230583 |
dvossel | 2009-11-20 11:26:20 -0600 (Fri, 20 Nov 2009) | 6 lines
audiohook signal trigger on every status change (issue #14618)
Review: https://reviewboard.asterisk.org/r/434/ ........
* apps/app_mixmonitor.c, /: Merged revisions 230509 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r230509 | dvossel | 2009-11-19 15:26:21 -0600
(Thu, 19 Nov 2009) | 17 lines Merged revisions 230508 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r230508 | dvossel | 2009-11-19 15:22:46 -0600 (Thu, 19 Nov 2009)
| 10 lines fixes MixMonitor thread not exiting when
StopMixMonitor is used (closes issue #16152) Reported by: AlexMS
Patches: stopmixmonitor_1.4.diff uploaded by dvossel (license
671) Tested by: dvossel, AlexMS Review:
https://reviewboard.asterisk.org/r/424/ ........ ................
2009-11-16 16:41 +0000 [r230250-230384] Kevin P. Fleming <kpfleming@digium.com>
* /, apps/app_fax.c: Merged revisions 230381 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r230381 |
kpfleming | 2009-11-16 10:40:25 -0600 (Mon, 16 Nov 2009) | 1 line
Fix another buglet in T.38 session teardown at the end of FAX
sessions. ........
* /, apps/app_fax.c: Merged revisions 230343 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r230343 |
kpfleming | 2009-11-16 06:51:59 -0600 (Mon, 16 Nov 2009) | 2
lines Ensure that only one end of a T.38 session initiates
teardown at completion. ........
* channels/chan_iax2.c, /: Merged revisions 230247 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r230247 | kpfleming | 2009-11-15 11:23:02 -0600
(Sun, 15 Nov 2009) | 12 lines Merged revisions 230246 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r230246 | kpfleming | 2009-11-15 11:19:06 -0600 (Sun, 15 Nov
2009) | 6 lines Correct mistaken option name in error message.
The configuration option for allowing hosts to make
non-token-based calls is 'calltokenoptional', not
'calltokenignore'. (reported on asterisk-users) ........
................
2009-11-13 22:01 +0000 [r229969-230148] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 230145 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r230145 | file | 2009-11-13 16:00:44 -0600 (Fri, 13 Nov 2009) |
15 lines Merged revisions 230144 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r230144 | file | 2009-11-13 16:00:19 -0600 (Fri, 13 Nov 2009) | 8
lines Respect the maddr parameter in the Via header. (closes
issue #14446) Reported by: frawd Patches: via_maddr.patch
uploaded by frawd (license 610) Tested by: frawd ........
................
* channels/chan_local.c, /: Merged revisions 230039 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r230039 | file | 2009-11-13 13:44:53 -0600 (Fri,
13 Nov 2009) | 16 lines Merged revisions 230038 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r230038 | file | 2009-11-13 13:44:07 -0600 (Fri, 13 Nov 2009) | 9
lines Fix a crash caused by two threads thinking they should both
free the chan_local private structure when only one should.
(closes issue #15314) Reported by: sroberts Patches:
Issue15314_Move_Nulling_owner.patch uploaded by davidw (license
780) Tested by: davidw, lottc ........ ................
* configs/extensions.conf.sample, /, apps/app_chanisavail.c: Merged
revisions 229966 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r229966 | file | 2009-11-13 11:20:26 -0600 (Fri, 13 Nov 2009) |
13 lines Merged revisions 229965 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r229965 | file | 2009-11-13 11:19:59 -0600 (Fri, 13 Nov 2009) | 6
lines Document a limitation in the AVAILSTATUS variable from
ChanIsAvail and provide a workaround for it that does not change
existing behavior. (closes issue #14426) Reported by: macli
........ ................
2009-11-13 Leif Madsen <lmadsen@digium.com>
* Release Asterisk 1.6.2.0-rc6
2009-11-13 15:57 +0000 [r229915] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 229912 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r229912 |
file | 2009-11-13 09:56:16 -0600 (Fri, 13 Nov 2009) | 2 lines Fix
T.38 negotiation regression introduced with the SDP parser
changes. ........
2009-11-12 23:31 +0000 [r229752] Jason Parker <jparker@digium.com>
* channels/chan_oss.c, /: Merged revisions 229750 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r229750 |
qwell | 2009-11-12 17:30:10 -0600 (Thu, 12 Nov 2009) | 1 line Fix
mute toggling on OSS channels. ........
2009-11-12 16:47 +0000 [r229671] David Vossel <dvossel@digium.com>
* funcs/func_audiohookinherit.c, /: Merged revisions 229670 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r229670 | dvossel | 2009-11-12 10:44:39 -0600
(Thu, 12 Nov 2009) | 12 lines Merged revisions 229669 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r229669 | dvossel | 2009-11-12 10:41:49 -0600 (Thu, 12 Nov 2009)
| 6 lines fixes merging error, datastore was being freed in the
wrong function. (closes issue #16219) Reported by: aragon
........ ................
2009-11-11 20:49 +0000 [r229570] David Ruggles <thedavidfactor@gmail.com>
* doc/externalivr.txt: Merged revisions 229568 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r229568 |
diruggles | 2009-11-11 15:47:06 -0500 (Wed, 11 Nov 2009) | 9
lines Remove non-functional feature from ExternalIVR
documentation Remove non-functional socket implementation of
ExternalIVR from documentation (closes issue #16225) Reported by:
thedavidfactor Patches: externalivr.txt.20091111.1542.patch
uploaded by thedavidfactor (license 903) ........
2009-11-11 19:56 +0000 [r229492-229502] David Brooks <dbrooks@digium.com>
* main/pbx.c, /: Merged revisions 229499 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r229499 | dbrooks | 2009-11-11 13:48:18 -0600 (Wed, 11 Nov 2009)
| 15 lines Merged revisions 229498 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r229498 | dbrooks | 2009-11-11 13:46:19 -0600 (Wed, 11 Nov 2009)
| 8 lines Solaris doesn't like NULL going to ast_log Solaris will
crash if NULL is passed to ast_log. This simple patch simply uses
S_OR to get around this. (closes issue #15392) Reported by:
yrashk ........ ................
* /, apps/app_softhangup.c: Merged revisions 229460 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r229460 | dbrooks | 2009-11-11 12:13:56 -0600 (Wed, 11 Nov 2009)
| 7 lines Flags not initialized in app_softhangup.c, causing
undefined behavior Trivial patch [kobaz] to initialize an
ast_flags = {0} (closes issue #16129) Reported by: kobaz ........
2009-11-10 22:17 +0000 [r229366] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c, /: Merged revisions 229361 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r229361 | tilghman | 2009-11-10 16:14:22 -0600 (Tue, 10 Nov 2009)
| 19 lines Merged revisions 229360 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r229360 | tilghman | 2009-11-10 16:09:16 -0600 (Tue, 10 Nov 2009)
| 12 lines If two pattern classes start with the same digit and
have the same number of characters, they will compare equal. The
example given in the issue report is that of [234] and [246],
which have these characteristics, yet they are clearly not
equivalent. The code still uses these two characteristics, yet
when the two scores compare equal, an additional check will be
done to compare all characters within the class to verify
equality. (closes issue #15421) Reported by: jsmith Patches:
20091109__issue15421__2.diff.txt uploaded by tilghman (license
14) Tested by: jsmith, thedavidfactor ........ ................
2009-11-10 22:04 +0000 [r229359] David Ruggles <thedavidfactor@gmail.com>
* doc/externalivr.txt: Merged revisions 229356 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r229356 | diruggles | 2009-11-10 17:01:50 -0500 (Tue, 10 Nov
2009) | 16 lines Merged revisions 229355 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r229355 | diruggles | 2009-11-10 16:45:15 -0500 (Tue, 10 Nov
2009) | 9 lines Fix ExternalIVR Documentation Remove
documentation for event that doesn't function (closes issue
#16220) Reported by: thedavidfactor Patches:
externalivr.txt.20091110.1622.patch uploaded by thedavidfactor
(license 903) ........ ................
2009-11-10 21:33 +0000 [r229354] Tilghman Lesher <tlesher@digium.com>
* apps/app_stack.c, /: Merged revisions 229351 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r229351 |
tilghman | 2009-11-10 15:22:50 -0600 (Tue, 10 Nov 2009) | 7 lines
When GOSUB is invoked within an AGI, it may not exit correctly.
(closes issue #16216) Reported by: atis Patches:
20091110__atis_work.diff.txt uploaded by tilghman (license 14)
Tested by: atis ........
2009-11-10 20:09 +0000 [r229285] Joshua Colp <jcolp@digium.com>
* /, codecs/codec_g726.c: Merged revisions 229282 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r229282 | file | 2009-11-10 16:06:13 -0400 (Tue, 10 Nov 2009) |
15 lines Merged revisions 229281 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r229281 | file | 2009-11-10 16:03:14 -0400 (Tue, 10 Nov 2009) | 8
lines Remove broken support for direct transcoding between G.726
RFC3551 and G.726 AAL2. On some systems the translation core
would actually consider g726aal2 -> g726 -> signed linear to be a
quicker path then g726aal2 -> signed linear which exposed this
problem. (closes issue #15504) Reported by: globalnetinc ........
................
2009-11-10 17:52 +0000 [r229232] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c, /: Merged revisions 229168 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r229168 | dvossel | 2009-11-10 11:16:49 -0600
(Tue, 10 Nov 2009) | 15 lines Merged revisions 229167 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r229167 | dvossel | 2009-11-10 11:15:57 -0600 (Tue, 10 Nov 2009)
| 9 lines don't crash on log message in solaris AST-2009-006
(closes issue #16206) Reported by: bklang Tested by: bklang
........ ................
2009-11-10 17:39 +0000 [r229231] David Ruggles <thedavidfactor@gmail.com>
* doc/externalivr.txt: Merged revisions 229228 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r229228 | diruggles | 2009-11-10 12:33:47 -0500 (Tue, 10 Nov
2009) | 18 lines Merged revisions 229191 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r229191 | diruggles | 2009-11-10 12:23:59 -0500 (Tue, 10 Nov
2009) | 11 lines Document ExternalIVR event tag collision
ExternalIVR uses the D tag for two different event types. This
documents that behavior and how to differentiate between the two
cases. Also includes a minor spelling fix and clarification
(closes issue #16211) Reported by: thedavidfactor Patches:
externalivr.txt.20091109.1507.patch uploaded by thedavidfactor
(license 903) ........ ................
2009-11-10 15:47 +0000 [r229101] Matthew Nicholson <mnicholson@digium.com>
* UPGRADE-1.6.txt, main/editline/makelist.in, UPGRADE.txt: Reset
props that were accidently deleted in 229088.
2009-11-10 15:28 +0000 [r229094] David Vossel <dvossel@digium.com>
* res/res_config_pgsql.c, /: Merged revisions 229093 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r229093 | dvossel | 2009-11-10 09:27:45 -0600 (Tue, 10 Nov 2009)
| 11 lines fixes pgsql double free of threadstorage A thread
storage variable was being freed incorrectly, which resulted in a
double free if two queries were made in the same thread. (closes
issue #16011) Reported by: cristiandimache Patches:
issue16011.diff uploaded by dvossel (license 671) ........
2009-11-10 15:16 +0000 [r229088] Matthew Nicholson <mnicholson@digium.com>
* UPGRADE-1.6.txt, main/editline/makelist.in, channels/chan_sip.c,
UPGRADE.txt: Reverted revision 202007. (closes issue #16175)
Reported by: paul-tg Tested by: paul-tg
2009-11-10 11:25 +0000 [r229078] Gavin Henry <ghenry@suretecsystems.com>
* contrib/scripts/asterisk.ldap-schema, /: Merged revisions 229050
via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r229050 | ghenry | 2009-11-10 11:16:10 +0000 (Tue, 10
Nov 2009) | 20 lines Schema file additions * Added
AsteriskDialplan, AsteriskAccount and AsteriskMailbox
objectClasses to allow standalone dialplan, account and mailbox
entries (STRUCTURAL) * Added new Fields: - AstAccountLanguage,
AstAccountTransport, AstAccountPromiscRedir, -
AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
- AstAccountVideoSupport, AstAccountIgnoreSDPVersion * Removed
redundant IPaddr (there's already IPAddress) - Gives more
configuration Flags for SIP-Users available (tested) - Allows to
create Asterisk Attributes in defined Asterisk ObjectClasses
without extensibleObject (which really should be the last
resort); gives also additional possibilities for LDAP-filter
(closes issue #15874) Reported by: Medozas Patches:
asterisk.ldap-schema.patch uploaded by Medozas (license 41)
Tested by: Medozas, suretec ........
2009-11-09 22:59 +0000 [r229017] Terry Wilson <twilson@digium.com>
* channels/chan_local.c, /: Merged revisions 229015 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r229015 | twilson | 2009-11-09 16:50:22 -0600 (Mon, 09 Nov 2009)
| 8 lines Don't crash when bridge->tech_pvt == NULL This is a
similar solution to what is in place for chan_agent (closes issue
#16003) Reported by: atis Tested by: twilson ........
2009-11-09 22:17 +0000 [r229012] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: fixes segfault when transferring a queue
caller In sip_hangup we attempted to lock p->owner after we set
it to NULL. Thanks to fhackenberger for reporting the issue and
submitting a patch. (closes issue #15848) Reported by:
fhackenberger Patches: digium_bug_0015848 uploaded by
fhackenberger (license 592) Tested by: fhackenberger, lmadsen,
TomS, shin-shoryuken, dvossel
2009-11-09 Leif Madsen <lmadsen@digium.com>
* Release Asterisk 1.6.2.0-rc5
2009-11-09 15:40 +0000 [r228900] Leif Madsen <lmadsen@digium.com>
* main/channel.c: Merged revisions 228897 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r228897 | lmadsen | 2009-11-09 09:38:38 -0600 (Mon, 09 Nov 2009)
| 14 lines Merged revisions 228896 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r228896 | lmadsen | 2009-11-09 09:37:43 -0600 (Mon, 09 Nov 2009)
| 6 lines Update WARNING message. Update a WARNING message to
give a suggested fix when encountered. (closes issue #16198)
Reported by: atis Tested by: atis ........ ................
2009-11-09 14:48 +0000 [r228859] Matthew Nicholson <mnicholson@digium.com>
* /, include/asterisk/lock.h: Merged revisions 228858 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r228858 | mnicholson | 2009-11-09 08:37:07 -0600
(Mon, 09 Nov 2009) | 15 lines Merged revisions 228827 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r228827 | mnicholson | 2009-11-09 08:16:03 -0600 (Mon, 09 Nov
2009) | 8 lines Perform limited bounds checking when destroying
ast_mutex_t structures to make sure we don't try to use negative
indices. (closes issue #15588) Reported by: zerohalo Patches:
20090820__issue15588.diff.txt uploaded by tilghman (license 14)
Tested by: zerohalo ........ ................
2009-11-06 22:37 +0000 [r228694] David Vossel <dvossel@digium.com>
* main/channel.c, /: Merged revisions 228693 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r228693 | dvossel | 2009-11-06 16:35:44 -0600 (Fri, 06 Nov 2009)
| 16 lines Merged revisions 228692 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r228692 | dvossel | 2009-11-06 16:33:27 -0600 (Fri, 06 Nov 2009)
| 9 lines fixes audiohook write crash occuring in chan_spy
whisper mode. After writing to the audiohook list in ast_write(),
frames were being freed incorrectly. Under certain conditions
this resulted in a double free crash. (closes issue #16133)
Reported by: wetwired ........ ................
2009-11-06 20:26 +0000 [r228649] Matthew Nicholson <mnicholson@digium.com>
* funcs/func_base64.c, /, main/utils.c: Merged revisions 228620 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r228620 | mnicholson | 2009-11-06 13:47:11 -0600
(Fri, 06 Nov 2009) | 15 lines Merged revisions 228378 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r228378 | mnicholson | 2009-11-06 10:26:59 -0600 (Fri, 06 Nov
2009) | 8 lines Properly handle '=' while decoding base64
messages and null terminate strings returned from BASE64_DECODE.
(closes issue #15271) Reported by: chappell Patches:
base64_fix.patch uploaded by chappell (license 8) Tested by:
kobaz ........ ................
2009-11-06 18:43 +0000 [r228551] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 228548 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r228548 | file | 2009-11-06 14:37:59 -0400 (Fri, 06 Nov 2009) |
11 lines Merged revisions 228547 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r228547 | file | 2009-11-06 14:32:58 -0400 (Fri, 06 Nov 2009) | 4
lines Don't overwrite caller ID name on a trunk with the
configured fullname when using users.conf (issue ABE-1989)
........ ................
2009-11-06 Leif Madsen <lmadsen@digium.com>
* Release Asterisk 1.6.2.0-rc4
2009-11-06 17:54 +0000 [r228504] Joshua Colp <jcolp@digium.com>
* doc/tex/localchannel.tex, /: Merged revisions 228499 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r228499 | file | 2009-11-06 13:52:00 -0400 (Fri, 06 Nov 2009) | 2
lines Fix the localchannel.tex file. ........
2009-11-06 17:24 +0000 [r228421-228447] David Vossel <dvossel@digium.com>
* codecs/codec_ilbc.c, /: Merged revisions 228441 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r228441 |
dvossel | 2009-11-06 11:22:31 -0600 (Fri, 06 Nov 2009) | 3 lines
Fixes merging issue from 1.4, frame data is held in data.ptr in
trunk ........
* codecs/codec_ilbc.c, /: Merged revisions 228420 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r228420 | dvossel | 2009-11-06 11:09:01 -0600 (Fri, 06 Nov 2009)
| 19 lines Merged revisions 228418 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r228418 | dvossel | 2009-11-06 11:07:13 -0600 (Fri, 06 Nov 2009)
| 13 lines fixes segfault in iLBC For reasons not yet known, it
appears possible for an ast_frame to have a datalen greater than
zero while the actual data is NULL during Packet Loss
Concealment. Most codecs don't support PLC so this doesn't affect
them. This patch catches the malformed frame and prevents the
crash from occuring. Additional efforts to determine why it is
possible for a frame to look like this are still being
investigated. (issue #16979) ........ ................
2009-11-06 16:44 +0000 [r228413] Joshua Colp <jcolp@digium.com>
* /, main/abstract_jb.c: Merged revisions 228410 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r228410 | file | 2009-11-06 12:42:23 -0400 (Fri, 06 Nov 2009) |
14 lines Merged revisions 228409 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r228409 | file | 2009-11-06 12:41:20 -0400 (Fri, 06 Nov 2009) | 7
lines Fix a bug caused by a partially invalid frame (from the
jitterbuffer) passing through the Asterisk core. (closes issue
#15560) Reported by: jvandal (closes issue #15709) Reported by:
covici ........ ................
2009-11-06 15:43 +0000 [r228269-228340] David Vossel <dvossel@digium.com>
* /, main/astfd.c: Merged revisions 228339 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r228339 | dvossel | 2009-11-06 09:42:46 -0600 (Fri, 06 Nov 2009)
| 12 lines Merged revisions 228338 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r228338 | dvossel | 2009-11-06 09:41:41 -0600 (Fri, 06 Nov 2009)
| 5 lines fixes crash in astfd.c (closes issue #15981) Reported
by: slavon ........ ................
* funcs/func_audiohookinherit.c, /: Merged revisions 228268 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r228268 | dvossel | 2009-11-06 09:04:24 -0600 (Fri, 06
Nov 2009) | 9 lines fixes memory leak in func_audiohookinherit.c
(closes issue #15394) Reported by: boroda Patches:
bug15394_memoryleak_diff2.txt uploaded by dbrooks (license 790)
Tested by: dbrooks, boroda ........
2009-11-05 22:13 +0000 [r228198] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_meetme.c: Merged revisions 228196 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r228196 |
tilghman | 2009-11-05 16:12:45 -0600 (Thu, 05 Nov 2009) | 2 lines
Yet another error message in the dialplan (thanks,
rmudgett/russellb) ........
2009-11-05 21:27 +0000 [r228195] Jeff Peeler <jpeeler@digium.com>
* apps/app_chanspy.c, /: Merged revisions 228189 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r228189 |
jpeeler | 2009-11-05 15:23:06 -0600 (Thu, 05 Nov 2009) | 11 lines
Fix the fix for chanspy option o In 224178, I assumed the
uploaded patch was correct as it had received positive feedback.
The flags were being checked in the incorrect location. Upon
testing the fix this time it was also found that the flags from
the dialplan weren't being copied to the
chanspy_translation_helper. (closes issue #16167) Reported by:
marhbere ........
2009-11-05 21:27 +0000 [r228194] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_meetme.c: Merged revisions 228191 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r228191 |
tilghman | 2009-11-05 15:24:21 -0600 (Thu, 05 Nov 2009) | 7 lines
MEETME_INFO should not return a literal error message to the
dialplan. (closes issue #15450) Reported by: JimVanM Patches:
meetmeinfopatch.diff.txt uploaded by dbrooks (license 790) Tested
by: JimVanM ........
2009-11-05 19:42 +0000 [r228148] David Brooks <dbrooks@digium.com>
* channels/chan_misdn.c, /: Merged revisions 228145 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r228145 | dbrooks | 2009-11-05 13:34:50 -0600
(Thu, 05 Nov 2009) | 16 lines Merged revisions 228078 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r228078 | dbrooks | 2009-11-05 12:59:41 -0600 (Thu, 05 Nov 2009)
| 9 lines chan_misdn Asterisk 1.4.27-rc2 crash Crash related to
chan_misdn connection. Patch submitted by gknispel_proformatique,
tested by francesco_r. "I have many crash since i have upgraded
to Asterisk 1.4.27-rc2. Attached a full bt." This patch zeros out
an ast_frame. (closes issue #16041) Reported by: francesco_r
........ ................
2009-11-05 19:20 +0000 [r228093] Jason Parker <jparker@digium.com>
* channels/chan_vpb.cc, /: Merged revisions 228080 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r228080 | qwell | 2009-11-05 13:16:29 -0600
(Thu, 05 Nov 2009) | 15 lines Merged revisions 228079 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r228079 | qwell | 2009-11-05 13:14:25 -0600 (Thu, 05 Nov 2009) |
8 lines Fix crash on VPB exception when no hardware is present.
(closes issue #14970) Reported by: tzafrir Patches:
vpb_exception.diff uploaded by tzafrir (license 46) Tested by:
markwaters ........ ................
2009-11-05 17:14 +0000 [r228017] Tilghman Lesher <tlesher@digium.com>
* apps/app_externalivr.c, /: Merged revisions 228015 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r228015 | tilghman | 2009-11-05 11:08:02 -0600 (Thu, 05 Nov 2009)
| 4 lines Don't crash if no arguments are passed. (closes issue
#16119) Reported by: thedavidfactor ........
2009-11-04 23:53 +0000 [r227947] Jeff Peeler <jpeeler@digium.com>
* res/res_monitor.c, /: Merged revisions 227945 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r227945 | jpeeler | 2009-11-04 17:50:59 -0600 (Wed, 04 Nov 2009)
| 21 lines Merged revisions 227944 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r227944 | jpeeler | 2009-11-04 17:47:08 -0600 (Wed, 04 Nov 2009)
| 14 lines Fix incorrect filename comparsion after monitor file
change The logic to detect if a requested file is indeed a
different file from the current file was incorrect. The main
issue being confusion of the use of filename_base which was
previously set without pathing information and then compared to
another full path. Robust file comparison logic has been added to
properly check if two files are the same even if symlinks are
used. (closes issue #15313) Reported by: caspy Patches:
20091103__issue15313__1.4.diff.txt uploaded by jpeeler (license
325) but mostly tilghman's work ........ ................
2009-11-04 21:09 +0000 [r227760-227831] Matthew Nicholson <mnicholson@digium.com>
* apps/app_dial.c, /: Merged revisions 227829 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r227829 | mnicholson | 2009-11-04 15:03:33 -0600 (Wed, 04 Nov
2009) | 17 lines Merged revisions 227827 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r227827 | mnicholson | 2009-11-04 14:52:27 -0600 (Wed, 04 Nov
2009) | 10 lines This patch modifies the Dial application to
monitor the calling channel for hangups while playing back
announcements. (closes issue #16005) Reported by: falves11
Patches: dial-announce-hangup-fix1.diff uploaded by mnicholson
(license 96) Tested by: mnicholson, falves11 Review:
https://reviewboard.asterisk.org/r/407/ ........ ................
* channels/chan_sip.c: Modify the SDP parsing code to parse session
and media level items separately. With the new code, media level
proprieties should no longer be confused with session level
proprieties. This change also reorganizes some of the SDP parsing
code which should make it easier to manage in the future. (closes
issue #14994) Reported by: frawd
2009-11-04 19:28 +0000 [r227733-227748] Joshua Colp <jcolp@digium.com>
* /, static-http/prototype.js: Merged revisions 227739 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r227739 | file | 2009-11-04 15:26:19 -0400 (Wed,
04 Nov 2009) | 12 lines Merged revisions 227735 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r227735 | file | 2009-11-04 15:25:37 -0400 (Wed, 04 Nov 2009) | 5
lines Fix a security issue where it may be possible for someone
to execute a cross-site AJAX request exploit. (AST-2009-009)
........ ................
* /, channels/chan_sip.c: Merged revisions 227712 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r227712 | file | 2009-11-04 15:20:46 -0400 (Wed, 04 Nov 2009) |
12 lines Merged revisions 227700 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r227700 | file | 2009-11-04 15:17:39 -0400 (Wed, 04 Nov 2009) | 5
lines Fix a security issue where sending a REGISTER with a
differing username in the From URI and Authorization header would
reveal whether it was valid or not. (AST-2009-008) ........
................
2009-11-03 20:01 +0000 [r227375] Jason Parker <jparker@digium.com>
* Makefile, /, main/Makefile: Merged revisions 227372 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r227372 | qwell | 2009-11-03 13:59:46 -0600 (Tue, 03 Nov 2009) |
9 lines Fix some build issues on Solaris. (closes issue #14517)
(SWP-109) Reported by: asgaroth Patches: bug_14517.diff uploaded
by snuffy (license 35) Tested by: asgaroth, snuffy, dougm, qwell
........
2009-11-03 19:49 +0000 [r227364-227371] Leif Madsen <lmadsen@digium.com>
* apps/app_controlplayback.c, /: Merged revisions 227368 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r227368 | lmadsen | 2009-11-03 13:48:53 -0600 (Tue, 03
Nov 2009) | 8 lines Change warning message to debug message.
app_controlplayback outputs a warning, when in fact it is normal.
(closes issue #16071) Reported by: atis Patches:
controlplayback_warning.patch uploaded by atis (license 242)
........
* configs/extensions.conf.sample, /: Merged revisions 227361 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r227361 | lmadsen | 2009-11-03 13:25:18 -0600 (Tue, 03
Nov 2009) | 11 lines Additional fixes to the
extensions.conf.sample file. Update the extensions.conf.sample
[stdexten] context so that we use the variable instead of
requiring it to be passed explicitly. Also updated uses of the
[stdexten] context throughout. (closes issue #15858) Reported by:
pprindeville Patches: stdexten-context-update.txt uploaded by
lmadsen (license 10) Tested by: pprindeville ........
2009-11-03 18:15 +0000 [r227280] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: Merged revisions 227275 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r227275 | rmudgett | 2009-11-03 11:55:47 -0600 (Tue, 03 Nov 2009)
| 4 lines Make sure the outgoing flag is cleared if a new channel
fails to get created for outgoing calls. This is the relevant
portion of asterisk/trunk -r226648 ........
2009-11-03 17:14 +0000 [r227239] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 227238 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r227238 |
dvossel | 2009-11-03 11:12:52 -0600 (Tue, 03 Nov 2009) | 5 lines
user.conf entries in SIP were not having their peer type set.
(closes issue #16120) Reported by: jsmith ........
2009-11-03 15:40 +0000 [r227170] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 227167 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r227167 | file | 2009-11-03 11:37:08 -0400 (Tue, 03 Nov 2009) |
12 lines Merged revisions 227166 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r227166 | file | 2009-11-03 11:36:16 -0400 (Tue, 03 Nov 2009) | 5
lines Fix a bug where an RPID header could be generated with a
blank username in the URI. (closes issue #15909) Reported by:
kobaz ........ ................
2009-11-03 15:25 +0000 [r227165] Leif Madsen <lmadsen@digium.com>
* configs/extensions.conf.sample, /: Merged revisions 227162 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r227162 | lmadsen | 2009-11-03 09:19:47 -0600 (Tue, 03
Nov 2009) | 7 lines Update extensions.conf.sample file to fix
incorrect extensions. (closes issue #15857) Reported by:
pprindeville Patches: stdexten.patch#2 uploaded by pprindeville
(license 347) Tested by: pprindeville ........
2009-11-03 13:51 +0000 [r227156] Olle Johansson <oej@edvina.net>
* Makefile, /, channels/chan_sip.c: Merged revisions 227091 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r227091 | oej | 2009-11-03 12:11:15 +0100 (Tis,
03 Nov 2009) | 15 lines Merged revisions 227088 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r227088 | oej | 2009-11-03 11:29:59 +0100 (Tis, 03 Nov 2009) | 7
lines Use proper response code when violating Contact ACL's.
https://reviewboard.asterisk.org/r/415/ Thanks kpfleming for a
quick review. (EDVX-003) ........ ................
2009-11-02 21:06 +0000 [r226978] David Brooks <dbrooks@digium.com>
* channels/chan_sip.c: SIP channel name uniqueness SIP channel
names were supposed to be unique by way of a name suffix derived
from the pointer to the channel's private data. Uniqueness was
preserved on 32-bit systems, but not on 64-bit systems. This
patch, as suggested by kpfleming, replaces this suffix with a
simple incremented unsigned int. (closes issue #15152) Reported
by: palbrecht Review: https://reviewboard.asterisk.org/r/420/
2009-11-02 18:12 +0000 [r226893] Joshua Colp <jcolp@digium.com>
* apps/app_dial.c, /: Merged revisions 226890 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r226890 | file | 2009-11-02 14:08:54 -0400 (Mon, 02 Nov 2009) |
18 lines Merged revisions 226889 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r226889 | file | 2009-11-02 14:08:11 -0400 (Mon, 02 Nov 2009) |
11 lines Fix a bug where the recorded privacy introduction file
would not get removed if the caller hung up while the called
party had not yet answered. This was fixed by introducing an
argument to the 'n' option which, when enabled, removes the
introduction file under all scenarios. This was done to preserve
the behavior that has existed for quite some time. (closes issue
#14674) Reported by: ulogic Patches: bug14674.patch uploaded by
jpeeler (license 325) ........ ................
2009-11-02 17:17 +0000 [r226815] Tilghman Lesher <tlesher@digium.com>
* /, contrib/init.d/rc.redhat.asterisk: Merged revisions 226812 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r226812 | tilghman | 2009-11-02 11:15:31 -0600
(Mon, 02 Nov 2009) | 15 lines Merged revisions 226811 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r226811 | tilghman | 2009-11-02 11:14:20 -0600 (Mon, 02 Nov 2009)
| 8 lines Don't allow two separate instances of safe_asterisk
when restarting from the init script. (closes issue #14562)
Reported by: davidw Patches: Initially
20091022__issue14562.diff.txt uploaded by tilghman (license 14)
Modified to 20091030__Issue14562_diff.txt uploaded by davidw
(license 780) Tested by: davidw ........ ................
2009-10-29 18:18 +0000 [r226540] Joshua Colp <jcolp@digium.com>
* doc/tex/localchannel.tex, channels/chan_local.c, /: Merged
revisions 226532 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r226532 | file | 2009-10-29 15:13:42 -0300 (Thu, 29 Oct 2009) |
13 lines Merged revisions 226531 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r226531 | file | 2009-10-29 15:11:26 -0300 (Thu, 29 Oct 2009) | 6
lines Add an option to enabling passing music on hold start and
stop requests through instead of acting on them in chan_local.
(closes issue #14709) Reported by: dimas ........
................
2009-10-28 21:32 +0000 [r226486] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* build_tools/get_documentation, /: remove empty awk pattern (//)
Solaris 10 nawk doesn't like the empty pattern such as '//' for
'always'. Just remove that. No pattern at all always matches.
Merged revisions 226453 via svnmerge from
http://svn.digium.com/svn/asterisk/trunk
2009-10-28 20:13 +0000 [r226379-226385] Leif Madsen <lmadsen@digium.com>
* configs/sip.conf.sample: Merged revisions 226384 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r226384 | lmadsen | 2009-10-28 15:11:07 -0500
(Wed, 28 Oct 2009) | 17 lines Merged revisions 226382 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r226382 | lmadsen | 2009-10-28 15:06:13 -0500 (Wed, 28 Oct 2009)
| 9 lines Update documentation in sip.conf.sample. Update the
documentation in sip.conf.sample in order to make it more clear
that directmedia/canreinvite do not cause Asterisk to ignore
reINVITEs. It is only used to stop Asterisk from generating a
reINVITE, but does not stop it from accepting them if necessary.
(closes issue #15644) Reported by: lmadsen ........
................
* doc/tex/channelvariables.tex: Merged revisions 226378 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r226378 | lmadsen | 2009-10-28 14:50:00 -0500
(Wed, 28 Oct 2009) | 15 lines Merged revisions 226377 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r226377 | lmadsen | 2009-10-28 14:48:29 -0500 (Wed, 28 Oct 2009)
| 7 lines Update CALLINGSUBADDR channel variable documentation.
(closes issue #15734) Reported by: alecdavis Patches:
channelvariables.tex.diff.txt uploaded by alecdavis (license 585)
Tested by: alecdavis ........ ................
2009-10-28 18:06 +0000 [r226170-226308] Tilghman Lesher <tlesher@digium.com>
* /, include/asterisk/linkedlists.h: Merged revisions 226305 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r226305 | tilghman | 2009-10-28 13:04:05 -0500
(Wed, 28 Oct 2009) | 9 lines Merged revisions 226304 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r226304 | tilghman | 2009-10-28 13:02:25 -0500 (Wed, 28
Oct 2009) | 2 lines Fix documentation (pointed out by
TheDavidFactor on #-dev) ........ ................
* main/manager.c, /: Merged revisions 226159 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r226159 | tilghman | 2009-10-27 15:22:07 -0500 (Tue, 27 Oct 2009)
| 14 lines Merged revisions 226138 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r226138 | tilghman | 2009-10-27 15:16:49 -0500 (Tue, 27 Oct 2009)
| 7 lines Manager output is not always NULL-terminated, so force
a NULL at the end of the filestream. (closes issue #15495)
Reported by: pdf Patches: 20090916__issue15495.diff.txt uploaded
by tilghman (license 14) Tested by: pdf ........ ................
2009-10-27 17:12 +0000 [r226101] Terry Wilson <twilson@digium.com>
* res/res_http_post.c, /: Merged revisions 226099 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r226099 |
twilson | 2009-10-27 11:48:54 -0500 (Tue, 27 Oct 2009) | 2 lines
Don't prepend the URI prefix to the post directory ........
2009-10-27 00:16 +0000 [r226055] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* /, configure, configure.ac: detect ARM Linux EABI OSARCH as
linux-gnu instead of linux-gnueabi * Set OSARCH to linux-gnu even
if host_os is linux-gnueabi * When checking if we are Linux,
check OSARCH rather than host_os The newer ARM ABI ("EABI") shows
the OS name 'linux-gnueabi' rather than 'linux-gnu' . This patch
sets OSARCH to be 'linux-gnu' even in such a case. OSARCH is
tested for the value of 'linux-gnu' in one or two places in the
tree. This patch also fixes the check libcap to check for $OSARCH
rather than $host_os . See also:
http://wiki.debian.org/ArmEabiPort Merged revisions 225957 via
svnmerge from http://svn.digium.com/svn/asterisk/branches/1.4
Merged revisions 226018 via svnmerge from
http://svn.digium.com/svn/asterisk/trunk
2009-10-26 19:42 +0000 [r225914] Jeff Peeler <jpeeler@digium.com>
* /, channels/chan_sip.c: Merged revisions 225912 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r225912 |
jpeeler | 2009-10-26 14:40:26 -0500 (Mon, 26 Oct 2009) | 12 lines
ACL check not present for verifying SIP INVITEs The ACL check in
check_peer_ok was missing and has now been restored. The missing
check allowed for calls to be made on prohibited networks where
an ACL was defined in sip.conf and the allowguest option was set
to off. See the AST security advisory below for more information.
Merge code associated with AST-2009-007. (closes issue #16091)
Reported by: thom4fun ........
2009-10-26 15:56 +0000 [r225871] Kevin P. Fleming <kpfleming@digium.com>
* apps/app_fax.c: Backport audio handling loop fixes from trunk
version of app_fax. This backport resolves some issues handling
audio frames during FAX processing, and ensures that the FAX
application doesn't accidentally get notified of a T.38
switchover at the end of a successful FAX. (closes issue #16127)
2009-10-23 14:46 +0000 [r225651] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 225650 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r225650 |
dvossel | 2009-10-23 09:41:50 -0500 (Fri, 23 Oct 2009) | 3 lines
Fixes an iterator memory leak and uninitialized memory ........
2009-10-23 14:08 +0000 [r225585] Kevin P. Fleming <kpfleming@digium.com>
* Makefile, /: Merged revisions 225582 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r225582 | kpfleming | 2009-10-23 09:02:42 -0500 (Fri, 23 Oct
2009) | 17 lines Merged revisions 225581 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r225581 | kpfleming | 2009-10-23 09:00:01 -0500 (Fri, 23 Oct
2009) | 10 lines Don't force menuselect.makeopts to be rebuilt on
every build. For some reason the menuselect.makeopts file was
listed as PHONY in the Makefile, resulting in 'make' needing to
rebuild it for every build. This then resulted in the embedded
module rules being rebuilt on every build, which can be slow and
is unnecessary. This patch fixes the problem by properly allowing
'make' to know when the menuselect.makeopts file needs to be
rebuilt (defining the proper dependencies). ........
................
2009-10-22 22:24 +0000 [r225516] Leif Madsen <lmadsen@digium.com>
* README, /: Merged revisions 225515 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r225515 |
lmadsen | 2009-10-22 17:24:03 -0500 (Thu, 22 Oct 2009) | 8 lines
Update README documentation. Update the README documentation to
correctly describe which CLI command you should use when
attempting to get help from the CLI. (closes issue #16064)
Reported by: thedavidfactor Patches: readme.patch uploaded by
thedavidfactor (license 903) ........
2009-10-22 21:55 +0000 [r225489] David Vossel <dvossel@digium.com>
* apps/app_externalivr.c, include/asterisk/tcptls.h, main/tcptls.c,
/, channels/chan_sip.c: Merged revisions 225445 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r225445 |
dvossel | 2009-10-22 14:55:51 -0500 (Thu, 22 Oct 2009) | 50 lines
SIP TCP/TLS: move client connection setup/write into tcp helper
thread, various related locking/memory fixes. What this patch
fixes 1.Moves sip TCP/TLS connection setup into the TCP helper
thread: Connection setup takes awhile and before this it was
being done while holding the monitor lock. 2.Moves TCP/TLS
writing to the TCP helper thread: Through the use of a packet
queue and an alert pipe, the TCP helper thread can now be woken
up to write data as well as read data. 3.Locking error: sip_xmit
returned an XMIT_ERROR without giving up the tcptls_session lock.
This lock has been completely removed from sip_xmit and placed in
the new sip_tcptls_write() function. 4.Memory leak: When creating
a tcptls_client the tls_cfg was alloced but never freed unless
the tcptls_session failed to start. Now the session_args for a
sip client are an ao2 object which frees the tls_cfg on
destruction. 5.Pointer to stack variable: During
sip_prepare_socket the creation of a client's
ast_tcptls_session_args was done on the stack and stored as a
pointer in the newly created tcptls_session. Depending on the
events that followed, there was a slight possibility that pointer
could have been accessed after the stack returned. Given the new
changes, it is always accessed after the stack returns which is
why I found it. Notable code changes 1.I broke tcptls.c's
ast_tcptls_client_start() function into two functions. One for
creating and allocating the new tcptls_session, and a separate
one for starting and handling the new connection. This allowed me
to create the tcptls_session, launch the helper thread, and then
establish the connection within the helper thread. 2.Writes to a
tcptls_session are now done within the helper thread. This is
done by using an alert pipe to wake up the thread if new data
needs to be sent. The thread's sip_threadinfo object contains the
alert pipe as well as the packet queue. 3.Since the threadinfo
object contains the alert pipe, it must now be accessed outside
of the helper thread for every write (queuing of a packet). For
easy lookup, I moved the threadinfo objects from a linked list to
an ao2_container. (closes issue #13136) Reported by: pabelanger
Tested by: dvossel, whys (closes issue #15894) Reported by:
dvossel Tested by: dvossel Review:
https://reviewboard.asterisk.org/r/380/ ........
2009-10-22 21:54 +0000 [r225488] Leif Madsen <lmadsen@digium.com>
* doc/valgrind.txt, contrib/valgrind.supp (added): Merged revisions
225485 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r225485 | lmadsen | 2009-10-22 16:52:30 -0500 (Thu, 22 Oct 2009)
| 19 lines Merged revisions 225484 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r225484 | lmadsen | 2009-10-22 16:51:52 -0500 (Thu, 22 Oct 2009)
| 11 lines Clean valgrind output by suppressing false errors.
Update valgrind.txt documentation and add valgrind.supp file in
order to allow those who are creating valgrind output to have
less false errors in the logfile. (closes issue #16007) Reported
by: atis Patches: valgrind.txt.diff uploaded by atis (license
242) asterisk2.supp uploaded by atis (license 242) Tested by:
atis, amorsen ........ ................
2009-10-22 17:14 +0000 [r225363] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c, /, apps/app_meetme.c, include/asterisk/channel.h:
Merged revisions 225360 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r225360 | tilghman | 2009-10-22 12:11:23 -0500 (Thu, 22 Oct 2009)
| 11 lines Merged revisions 225105 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r225105 | tilghman | 2009-10-21 11:02:12 -0500 (Wed, 21 Oct 2009)
| 4 lines Fix documentation for ast_softhangup() and correct the
misuse thereof. (closes issue #16103) Reported by: majorbloodnok
........ ................
2009-10-21 22:00 +0000 [r225035-225308] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c, /: Merged revisions 225307 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r225307 | dvossel | 2009-10-21 16:58:46 -0500
(Wed, 21 Oct 2009) | 20 lines Merged revisions 225243 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r225243 | dvossel | 2009-10-21 15:58:08 -0500 (Wed, 21 Oct 2009)
| 13 lines IAX2: VNAK loop caused by signaling frames with no
destination call number It is possible for the PBX thread to
queue up signaling frames before a destination call number is
received. This can result in signaling frames being sent out with
no destination call number. Since recent versions of Asterisk
require accurate destination callnumbers for all Full Frames,
this can cause a VNAK loop to occur. To resolve this no signaling
frames are sent until a destination callnumber is received, and
destination call numbers are now only required for iax_pvt
matching when the frame is an ACK. Review:
https://reviewboard.asterisk.org/r/413/ ........ ................
* configs/sip.conf.sample, channels/chan_iax2.c,
configs/iax.conf.sample, /, channels/chan_sip.c: Merged revisions
225033 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r225033 | dvossel | 2009-10-21 09:39:10 -0500 (Wed, 21 Oct 2009)
| 27 lines Merged revisions 225032 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r225032 | dvossel | 2009-10-21 09:37:04 -0500 (Wed, 21 Oct 2009)
| 20 lines IAX/SIP shrinkcallerid option The shrinking of caller
id removes '(', ' ', ')', non-trailing '.', and '-' from the
string. This means values such as 555.5555 and test-test result
in 555555 and testtest. There are instances, such as Skype
integration, where a specific value is passed via caller id that
must be preserved unmodified. This patch makes the shrinking of
caller id optional in chan_sip and chan_iax in order to support
such cases. By default this option is on to preserve previous
expected behavior. (closes issue #15940) Reported by: dimas
Patches: v2-15940.patch uploaded by dimas (license 88)
15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671)
Tested by: dvossel Review:
https://reviewboard.asterisk.org/r/408/ ........ ................
2009-10-20 22:11 +0000 [r224859] Tilghman Lesher <tlesher@digium.com>
* main/audiohook.c, funcs/func_speex.c, /: Merged revisions 224856
via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r224856 | tilghman | 2009-10-20 17:09:07 -0500
(Tue, 20 Oct 2009) | 12 lines Merged revisions 224855 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r224855 | tilghman | 2009-10-20 17:07:11 -0500 (Tue, 20 Oct 2009)
| 5 lines Pay attention to the return value of the manipulate
function. While this looks like an optimization, it prevents a
crash from occurring when used with certain audiohook callbacks
(diagnosed with SVN trunk, backported to 1.4 to keep the source
consistent across versions). ........ ................
2009-10-20 17:50 +0000 [r224777] Joshua Colp <jcolp@digium.com>
* /, main/features.c: Merged revisions 224774 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r224774 | file | 2009-10-20 14:47:34 -0300 (Tue, 20 Oct 2009) |
12 lines Merged revisions 224773 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r224773 | file | 2009-10-20 14:46:37 -0300 (Tue, 20 Oct 2009) | 5
lines Add support for relaying early media in the features
attended transfer option. (closes issue #14828) Reported by:
licedey ........ ................
2009-10-20 00:00 +0000 [r224674] Kevin P. Fleming <kpfleming@digium.com>
* main/rtp.c, /: Merged revisions 224671 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r224671 | kpfleming | 2009-10-19 18:47:39 -0500 (Mon, 19 Oct
2009) | 14 lines Merged revisions 224670 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r224670 | kpfleming | 2009-10-19 18:44:07 -0500 (Mon, 19 Oct
2009) | 7 lines Correct timestamp calculations when RTP sample
rates over 8kHz are used. While testing some endpoints that
support 16kHz and 32kHz sample rates, some log messages were
generated due to calc_rxstamp() computing timestamps in a way
that produced odd results, so this patch sanitizes the result of
the computations. ........ ................
2009-10-19 19:54 +0000 [r224571] Joshua Colp <jcolp@digium.com>
* apps/app_dial.c, /: Merged revisions 224567 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r224567 | file | 2009-10-19 16:49:09 -0300 (Mon, 19 Oct 2009) |
12 lines Merged revisions 224565 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r224565 | file | 2009-10-19 16:47:50 -0300 (Mon, 19 Oct 2009) | 5
lines Do not attempt early media bridging (ie: direct RTP setup)
if options are enabled that should prevent it. (closes issue
#14763) Reported by: cupotka ........ ................
2009-10-19 19:41 +0000 [r224563] Kevin P. Fleming <kpfleming@digium.com>
* formats/format_siren14.c, /: Merged revisions 224562 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r224562 | kpfleming | 2009-10-19 14:40:26 -0500 (Mon, 19 Oct
2009) | 1 line Remove useless debugging message. ........
2009-10-19 00:13 +0000 [r224447-224451] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c, /: Merged revisions 224448 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r224448 | tilghman | 2009-10-18 19:05:56 -0500 (Sun, 18 Oct 2009)
| 3 lines Allow ODBC storage to be queried with multiple
mailboxes, and remove multiple goto's. This corrects an issue
reported on the -users list. ........
* configs/res_odbc.conf.sample, /: Merged revisions 224446 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r224446 | tilghman | 2009-10-18 18:41:30 -0500 (Sun, 18
Oct 2009) | 2 lines Clarify that "forcecommit" is NOT an alias
for "autocommit", but instead controls the default disposition of
uncommitted transactions. ........
2009-10-17 01:58 +0000 [r224334] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 224331 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r224331 | jpeeler | 2009-10-16 20:36:08 -0500
(Fri, 16 Oct 2009) | 20 lines Merged revisions 224330 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r224330 | jpeeler | 2009-10-16 20:32:47 -0500 (Fri, 16 Oct 2009)
| 13 lines Fix stale caller id data from being reported in AMI
NewChannel event The problem here is that chan_dahdi is designed
in such a way to set certain values in the dahdi_pvt only once.
One of those such values is the configured caller id data in
chan_dahdi.conf. For PRI, the configured caller id data could be
overwritten during a call. Instead of saving the data and
restoring, it was decided that for all non-analog channels it was
simply best to not set the configured caller id in the first
place and also clear it at the end of the call. (closes issue
#15883) Reported by: jsmith ........ ................
2009-10-16 20:58 +0000 [r224264] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 224261 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r224261 | rmudgett | 2009-10-16 15:40:57 -0500
(Fri, 16 Oct 2009) | 25 lines Merged revisions 224260 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r224260 | rmudgett | 2009-10-16 15:25:23 -0500 (Fri, 16 Oct 2009)
| 18 lines Never released PRI channels when using Busy() or
Congestion() dialplan apps. When the Busy() or Congestion()
application is used towards ISDN (an ISDN progress is sent), the
responding ISDN Disconnect or Release may contain the ISDN cause
user busy or one of the congestion causes. In chan_dahdi.c these
causes will only set the needbusy or needcongestion flags and not
activate the softhangup procedure. Unfortunately only the latter
can interrupt the endless wait loop of Busy()/Congestion().
Result: PRI channels staying in state busy for the rest of
asterisk life or until the other end times out and forces the
call to clear. (in issue 0014292) Reported by: tomaso Patches:
disc_rel_userbusy.patch uploaded by tomaso (license 564) (This
patch is unrelated to the issue.) ........ ................
2009-10-15 15:58 +0000 [r224181] Jeff Peeler <jpeeler@digium.com>
* apps/app_chanspy.c, /: Merged revisions 224178 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r224178 |
jpeeler | 2009-10-15 10:57:14 -0500 (Thu, 15 Oct 2009) | 11 lines
Readd removed ability to allow listening to one side of the call
in app_chanspy (Option o) (closes issue #15675) Reported by:
john8675309 Patches: issue15675patchtrunk.txt uploaded by dbrooks
(license 790) Tested by: jgutierrez on users list:
http://lists.digium.com/pipermail/asterisk-users/2009-October/239155.html
........
2009-10-12 23:55 +0000 [r223835] Jeff Peeler <jpeeler@digium.com>
* apps/app_dial.c, /: Merged revisions 223832 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r223832 | jpeeler | 2009-10-12 18:48:09 -0500 (Mon, 12 Oct 2009)
| 15 lines Merged revisions 223804 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r223804 | jpeeler | 2009-10-12 18:12:50 -0500 (Mon, 12 Oct 2009)
| 8 lines Ensure ringing continues for branched calls after
progress is received While waiting for an answer, don't send
progress for branched calls for which ringing was sent. (closes
issue #15028) Reported by: fnordian ........ ................
2009-10-12 21:01 +0000 [r223757] David Vossel <dvossel@digium.com>
* configs/iax.conf.sample, /: Merged revisions 223756 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r223756 | dvossel | 2009-10-12 15:58:27 -0500 (Mon, 12 Oct 2009)
| 5 lines Clarifies trunkmaxsize, trunkfreq, and trunkmtu iax2
options SWP-151 ........
2009-10-12 14:37 +0000 [r223655] Kevin P. Fleming <kpfleming@digium.com>
* /, channels/chan_sip.c, apps/app_fax.c: Merged revisions 223652
via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r223652 | kpfleming | 2009-10-12 09:25:29 -0500 (Mon, 12
Oct 2009) | 13 lines Remove automatic switching from T.38 to
voice mode in chan_sip. chan_sip has some code to automatically
switch from T.38 mode to voice mode when a voice frame is written
to the channel while it is in T.38 mode; this was intended to
handle the situation when a FAX transmission has ended and the
channel is not yet hung up, but is causing problems at the
beginning of FAX sessions as well when there are still voice
frames 'in flight' at the time the T.38 negotiation completes.
This patch removes the automatic switchover, and changes app_fax
to explicitly switch off T.38 mode when the FAX transmission
process ends. (closes issue #16025) Reported by: jamicque
........
2009-10-11 17:32 +0000 [r223490] Russell Bryant <russell@digium.com>
* main/autoservice.c, /: Merged revisions 223487 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r223487 | russell | 2009-10-11 12:25:42 -0500 (Sun, 11 Oct 2009)
| 17 lines Merged revisions 223485-223486 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r223485 | russell | 2009-10-11 12:22:52 -0500 (Sun, 11 Oct 2009)
| 6 lines Don't use data outside of its scope. The purpose of
this code was to have a hangup frame put on the list of deferred
frames. However, the code that read the hangup frame was outside
of the scope of where the hangup frame was declared. ........
r223486 | russell | 2009-10-11 12:25:06 -0500 (Sun, 11 Oct 2009)
| 2 lines Remove some unnecessary code. ........ ................
2009-10-09 23:12 +0000 [r223406] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, channels/chan_h323.c: Fix interpretation
of PRIREDIRECTIONREASON set by chan_sip. This commit is the
simplest way to solve a problem that has already been solved in
trunk with the "COLP/CONP and Redirecting party information into
Asterisk" commit. In trunk the redirection reason is translated
into a generic redirect reason. I would have had to do the same
fix except chan_sip never reads PRIREDIRECTREASON. So both
chan_dahdi and chan_h323 have been modified to interpret the one
different redirect reason of "no-answer" properly and set the
ISDN reason code 2 of "no reply". (closes issue #15033) Reported
by: steinwej
2009-10-09 21:01 +0000 [r223333] Kevin P. Fleming <kpfleming@digium.com>
* /, apps/app_fax.c: Merged revisions 223330 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r223330 |
kpfleming | 2009-10-09 15:58:44 -0500 (Fri, 09 Oct 2009) | 10
lines Initiate T.38 switchover when acting as called party,
regardless of FAX direction. SendFAX() and ReceiveFAX() can be
given options to indicate whether they should act as the calling
or called party; this mode should be used to decide whether to
initiate a switchover to T.38, not the direction that the FAX
transfer will take place. (closes issue #16039) Reported by:
jamicque ........
2009-10-09 18:53 +0000 [r223286] Matthew Nicholson <mnicholson@digium.com>
* main/channel.c, /: Merged revisions 223273 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r223273 | mnicholson | 2009-10-09 13:34:08 -0500 (Fri, 09 Oct
2009) | 14 lines Merged revisions 223225 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r223225 | mnicholson | 2009-10-09 13:20:11 -0500 (Fri, 09 Oct
2009) | 8 lines Signal timeouts by returning AST_CONTROL_RINGING
when originating calls. (closes issue #15104) Reported by:
nblasgen Patches: manager-timeout1.diff uploaded by mnicholson
(license 96) Tested by: nblasgen, mnicholson ........
................
2009-10-09 18:29 +0000 [r223257] Mark Michelson <mmichelson@digium.com>
* apps/app_dial.c, /: Merged revisions 223215 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r223215 | mmichelson | 2009-10-09 13:17:34 -0500 (Fri, 09 Oct
2009) | 9 lines Recorded merge of revisions 223213 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r223213 | mmichelson | 2009-10-09 13:17:12 -0500 (Fri,
09 Oct 2009) | 3 lines Fix potential memory leak in app_dial.c
........ ................
2009-10-09 17:55 +0000 [r223208] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 223206 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r223206 | dvossel | 2009-10-09 12:53:37 -0500 (Fri, 09 Oct 2009)
| 16 lines Merged revisions 223205 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r223205 | dvossel | 2009-10-09 12:52:35 -0500 (Fri, 09 Oct 2009)
| 10 lines fixes sip registration using authuser in user.conf
(closes issue #14954) Reported by: tornblad Tested by:
mmichelson, tornblad, dvossel ........ ................
2009-10-09 17:27 +0000 [r223173] Matthew Nicholson <mnicholson@digium.com>
* cdr/cdr_sqlite3_custom.c, /: Merged revisions 223136 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r223136 | mnicholson | 2009-10-09 12:14:38 -0500 (Fri, 09 Oct
2009) | 8 lines Don't close the sqlite database when reloading.
Only close the database when unloading. (closes issue #15953)
Reported by: frawd Patches: sqlite3_rev220097.diff uploaded by
frawd (license 610) Tested by: frawd ........
2009-10-09 17:09 +0000 [r223089-223133] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 223132 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r223132 |
dvossel | 2009-10-09 11:54:02 -0500 (Fri, 09 Oct 2009) | 9 lines
'auth=' did not parse md5 secret correctly (closes issue #15949)
Reported by: ebroad Patches: authparsefix.patch uploaded by
ebroad (license 878) 15949_trunk.diff uploaded by dvossel
(license 671) Tested by: ebroad ........
* /, channels/chan_sip.c: Merged revisions 223088 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r223088 |
dvossel | 2009-10-09 10:49:30 -0500 (Fri, 09 Oct 2009) | 14 lines
p->peerauth is always empty in transmit_register() When using
callbackextension or specifing the peer name in a registration
string, the peer's specific auth settings set by the "auth="
strings within the peer definition are not used by the
registration. Thanks to ebroad for reporting the issue and
providing the patch. (closes issue #15955) Reported by: ebroad
Patches: regauthfix.patch uploaded by ebroad (license 878)
........
2009-10-08 20:00 +0000 [r222883] Russell Bryant <russell@digium.com>
* include/asterisk/frame.h, include/asterisk/file.h, main/frame.c,
/, main/file.c: Merged revisions 222880 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r222880 | russell | 2009-10-08 14:52:03 -0500 (Thu, 08 Oct 2009)
| 51 lines Merged revisions 222878 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r222878 | russell | 2009-10-08 14:45:47 -0500 (Thu, 08 Oct 2009)
| 44 lines Make filestream frame handling safer by isolating
frames before returning them. This patch is related to a number
of issues on the bug tracker that show crashes related to freeing
frames that came from a filestream. A number of fixes have been
made over time while trying to figure out these problems, but
there re still people seeing the crash. (Note that some of these
bug reports include information about other problems. I am
specifically addressing the filestream frame crash here.) I'm
still not clear on what the exact problem is. However, what is
_very_ clear is that we have seen quite a few problems over time
related to unexpected behavior when we try to use embedded frames
as an optimization. In some cases, this optimization doesn't
really provide much due to improvements made in other areas. In
this case, the patch modifies filestream handling such that the
embedded frame will not be returned. ast_frisolate() is used to
ensure that we end up with a completely mallocd frame. In
reality, though, we will not actually have to malloc every time.
For filestreams, the frame will almost always be allocated and
freed in the same thread. That means that the thread local frame
cache will be used. So, going this route doesn't hurt. With this
patch in place, some people have reported success in not seeing
the crash anymore. (SWP-150) (AST-208) (ABE-1834) (issue #15609)
Reported by: aragon Patches: filestream_frisolate-1.4.diff2.txt
uploaded by russell (license 2) Tested by: aragon, russell
(closes issue #15817) Reported by: zerohalo Tested by: zerohalo
(closes issue #15845) Reported by: marhbere Review:
https://reviewboard.asterisk.org/r/386/ ........ ................
2009-10-08 19:41 +0000 [r222874] David Vossel <dvossel@digium.com>
* main/netsock.c, /, include/asterisk/netsock.h: Merged revisions
222873 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r222873 |
dvossel | 2009-10-08 14:35:30 -0500 (Thu, 08 Oct 2009) | 6 lines
fixes an ast_netsock_list memory leak. ABE-1998 Review:
https://reviewboard.asterisk.org/r/395/ ........
2009-10-08 16:51 +0000 [r222695-222802] Richard Mudgett <rmudgett@digium.com>
* channels/misdn_config.c, /: Merged revisions 222799 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r222799 | rmudgett | 2009-10-08 11:44:33 -0500
(Thu, 08 Oct 2009) | 19 lines Merged revisions 222797 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r222797 | rmudgett | 2009-10-08 11:33:06 -0500 (Thu, 08 Oct 2009)
| 12 lines Fix memory leak if chan_misdn config parameter is
repeated. Memory leak when the same config option is set more
than once in an misdn.conf section. Why must this be considered?
Templates! Defining a template with default port options and
later adding to or overriding some of them. Patches:
memleak-misdn.patch JIRA ABE-1998 ........ ................
* channels/chan_misdn.c, /: Merged revisions 222692 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r222692 | rmudgett | 2009-10-07 16:56:36 -0500
(Wed, 07 Oct 2009) | 21 lines Merged revisions 222691 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r222691 | rmudgett | 2009-10-07 16:51:24 -0500 (Wed, 07 Oct 2009)
| 14 lines chan_misdn.c:process_ast_dsp() memory leak misdn.conf:
astdtmf must be set to "yes". With "no", buffer loss does not
occur. The translated frame "f2" when passing through
ast_dsp_process() is not freed whenever it is not used further in
process_ast_dsp(). Then in the end it is never ever freed.
Patches: translate.patch JIRA ABE-1993 ........ ................
2009-10-07 18:06 +0000 [r222549] Jason Parker <jparker@digium.com>
* /, configs/queues.conf.sample: Merged revisions 222548 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r222548 | qwell | 2009-10-07 13:04:56 -0500 (Wed, 07 Oct
2009) | 5 lines Remove 'keepstats' queue option from sample
config, as it's no longer used.
https://reviewboard.asterisk.org/r/115/ (closes issue #15820)
Reported by: kshumard ........
2009-10-07 18:00 +0000 [r222547] Sean Bright <sean@malleable.com>
* funcs/func_strings.c: Fix merge error.
2009-10-07 17:45 +0000 [r222544] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 222543 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r222543 | dvossel | 2009-10-07 12:44:52 -0500 (Wed, 07 Oct 2009)
| 14 lines Merged revisions 222542 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r222542 | dvossel | 2009-10-07 12:41:21 -0500 (Wed, 07 Oct 2009)
| 8 lines crash on transfer handle_invite_replaces() attempts to
uplock a pvt's owner channel without first verifing that it
exists. (issue #16027) ........ ................
2009-10-06 23:59 +0000 [r222354-222466] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 222463 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r222463 | jpeeler | 2009-10-06 18:56:01 -0500
(Tue, 06 Oct 2009) | 14 lines Merged revisions 222462 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r222462 | jpeeler | 2009-10-06 18:51:19 -0500 (Tue, 06 Oct 2009)
| 8 lines Add missing unlock(s) in dahdi_read (two cases in
trunk, and 1.6.2) (closes issue #15683) Reported by: alecdavis
........ ................
* channels/chan_dahdi.c: Fix potential crash when entire span
request is received. The variable index used in this scenario for
accessing the dahdi_pvts was wrong and was most likely copied
from the several other places it is used correctly. (closes issue
#15998) Reported by: tsearle Patches: dahdi_reset_crash.patch
uploaded by tsearle (license 373)
* channels/chan_dahdi.c, /: Merged revisions 222351 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r222351 | jpeeler | 2009-10-06 15:35:19 -0500 (Tue, 06 Oct 2009)
| 9 lines Fix 222298 (crash during destruction of second channel
when variable set with setvar). I mistakenly reasoned that setvar
would be used on all channels. Since it can be set per channel,
give each dahdi channel a copy of the variable. (related to
#15899) ........
2009-10-06 19:41 +0000 [r222311] Tilghman Lesher <tlesher@digium.com>
* cdr/cdr_pgsql.c, res/res_config_pgsql.c, /: Merged revisions
222309 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r222309 |
tilghman | 2009-10-06 14:31:39 -0500 (Tue, 06 Oct 2009) | 10
lines Change schema query to involve the use of an optional
schema parameter. This change is done in such a way as to allow
the driver to continue to function with older databases which
don't have these features. (closes issue #16000) Reported by:
jamicque Patches: 20091002__issue16000.diff.txt uploaded by
tilghman (license 14) 20091002__issue16000__1.6.1.diff.txt
uploaded by tilghman (license 14) Tested by: jamicque ........
2009-10-06 19:27 +0000 [r222304] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 222298 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r222298 | jpeeler | 2009-10-06 14:24:59 -0500 (Tue, 06 Oct 2009)
| 9 lines Fix crash during destruction of second channel when
variable set with setvar. The setvar line in chan_dahdi.conf is
shared among all the channels, so make sure to only free the
resources only when the last channel is destroyed. (closes issue
#15899) Reported by: tzafrir ........
2009-10-06 19:22 +0000 [r222289] Tilghman Lesher <tlesher@digium.com>
* res/ael/pval.c, /: Merged revisions 222273 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r222273 |
tilghman | 2009-10-06 14:17:11 -0500 (Tue, 06 Oct 2009) | 5 lines
When we call a gosub routine, the variables should be scoped to
avoid contaminating the caller. This affected the ~~EXTEN~~ hack,
where a subroutine might have changed the value before it was
used in the caller. Patch by myself, tested by ebroad on
#asterisk ........
2009-10-06 Leif Madsen <lmadsen@digium.com>
* Released Asterisk 1.6.2.0-rc3
2009-10-06 01:39 +0000 [r222113-222187] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_console.c, res/res_musiconhold.c, apps/app_queue.c,
channels/chan_iax2.c, main/astobj2.c, res/res_odbc.c,
res/res_clialiases.c, /, channels/chan_sip.c,
funcs/func_dialgroup.c, include/asterisk/astobj2.h,
res/res_phoneprov.c: Merged revisions 222176 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r222176 | kpfleming | 2009-10-05 20:24:24 -0500 (Mon, 05 Oct
2009) | 27 lines Recorded merge of revisions 222152 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05
Oct 2009) | 20 lines Fix ao2_iterator API to hold references to
containers being iterated. See Mantis issue for details of what
prompted this change. Additional notes: This patch changes the
ao2_iterator API in two ways: F_AO2I_DONTLOCK has become an enum
instead of a macro, with a name that fits our naming policy;
also, it is now necessary to call ao2_iterator_destroy() on any
iterator that has been created. Currently this only releases the
reference to the container being iterated, but in the future this
could also release other resources used by the iterator, if the
iterator implementation changes to use additional resources.
(closes issue #15987) Reported by: kpfleming Review:
https://reviewboard.asterisk.org/r/383/ ........ ................
* configs/sip.conf.sample, main/udptl.c, /, channels/chan_sip.c,
configs/udptl.conf.sample, UPGRADE.txt: Merged revisions 222110
via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r222110 | kpfleming | 2009-10-05 14:45:00 -0500 (Mon, 05
Oct 2009) | 25 lines Allow non-compliant T.38 endpoints to be
supportable via configuration option. Many T.38 endpoints
incorrectly send the maximum IFP frame size they can accept as
the T38FaxMaxDatagram value in their SDP, when in fact this value
is supposed to be the maximum UDPTL payload size (datagram size)
they can accept. If the value they supply is small enough (a
commonly supplied value is '72'), T.38 UDPTL transmissions will
likely fail completely because the UDPTL packets will not have
enough room for a primary IFP frame and the redundancy used for
error correction. If this occurs, the Asterisk UDPTL stack will
emit log messages warning that data loss may occur, and that the
value may need to be overridden. This patch extends the
't38pt_udptl' configuration option in sip.conf to allow the
administrator to override the value supplied by the remote
endpoint and supply a value that allows T.38 FAX transmissions to
be successful with that endpoint. In addition, in any SIP call
where the override takes effect, a debug message will be printed
to that effect. This patch also removes the T38FaxMaxDatagram
configuration option from udptl.conf.sample, since it has not
actually had any effect for a number of releases. In addition,
this patch cleans up the T.38 documentation in sip.conf.sample
(which incorrectly documented that T.38 support was passthrough
only). (issue #15586) Reported by: globalnetinc ........
2009-10-02 17:35 +0000 [r222032] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c, /: Merged revisions 222030 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r222030 | dvossel | 2009-10-02 12:34:07 -0500
(Fri, 02 Oct 2009) | 9 lines Merged revisions 222026 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r222026 | dvossel | 2009-10-02 12:32:13 -0500 (Fri, 02
Oct 2009) | 3 lines Removes unnecessary unlock, clarifies a
memcpy. ........ ................
2009-10-02 17:01 +0000 [r221923-221974] Tilghman Lesher <tlesher@digium.com>
* main/astobj2.c, /: Merged revisions 221971 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r221971 | tilghman | 2009-10-02 11:59:57 -0500 (Fri, 02 Oct 2009)
| 9 lines Merged revisions 221970 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r221970 | tilghman | 2009-10-02 11:58:03 -0500 (Fri, 02 Oct 2009)
| 2 lines Ensure the result of the hash function is positive.
Negative array offsets suck. ........ ................
* /, main/logger.c: Merged revisions 221920 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r221920 |
tilghman | 2009-10-01 22:04:34 -0500 (Thu, 01 Oct 2009) | 4 lines
Initialize a variable that we check immediately upon startup.
(closes issue #15973) Reported by: atis ........
2009-10-02 01:35 +0000 [r221879] Richard Mudgett <rmudgett@digium.com>
* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h, /:
Merged revisions 221844 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r221844 | rmudgett | 2009-10-01 20:09:31 -0500 (Thu, 01 Oct 2009)
| 33 lines Merged revisions 221769 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r221769 | rmudgett | 2009-10-01 18:18:28 -0500 (Thu, 01 Oct 2009)
| 26 lines Occasionally losing use of B channels in chan_misdn. I
have not been able to reproduce the problem of losing channels.
However, I have seen in the code a reentrancy problem that might
give these symptoms. The reentrancy patch does several things: 1)
Guards B channel and B channel structure allocation. 2) Makes the
B channel structure find routines more precise in locating
records. 3) Never leave a B channel allocated if we received
cause 44. The last item may cause temporary outgoing call
problems, but they should clear when the line becomes idle.
(closes issue #15490) Reported by: slutec18 Patches:
issue15490_channel_alloc_reentrancy.patch uploaded by rmudgett
(license 664) Tested by: rmudgett, slutec18 (closes issue #15458)
Reported by: FabienToune Patches:
issue15458_channel_alloc_reentrancy.patch uploaded by rmudgett
(license 664) Tested by: FabienToune, rmudgett, slutec18 ........
................
2009-10-02 00:07 +0000 [r221744-221780] Tilghman Lesher <tlesher@digium.com>
* main/asterisk.c, main/rtp.c, /, main/say.c: Merged revisions
221777 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r221777 | tilghman | 2009-10-01 18:59:15 -0500 (Thu, 01 Oct 2009)
| 9 lines Merged revisions 221776 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r221776 | tilghman | 2009-10-01 18:53:12 -0500 (Thu, 01 Oct 2009)
| 2 lines Fix a bunch of off-by-one errors ........
................
* /, channels/chan_sip.c: Merged revisions 221705 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r221705 |
tilghman | 2009-10-01 15:09:46 -0500 (Thu, 01 Oct 2009) | 2 lines
Revision 220906 (a merge from 1.4) was not merged correctly,
causing a problem with non-dynamic peers. ........
2009-10-01 19:35 +0000 [r221698] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 221697 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r221697 |
dvossel | 2009-10-01 14:33:33 -0500 (Thu, 01 Oct 2009) | 9 lines
outbound tls connections were not defaulting to port 5061 (closes
issue #15854) Reported by: dvossel Patches:
sip_port_config_trunk.diff uploaded by dvossel (license 671)
Tested by: dvossel ........
2009-10-01 16:57 +0000 [r221660] Matthew Nicholson <mnicholson@digium.com>
* /, channels/chan_sip.c: Merged revisions 221554,221589 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r221554 | oej | 2009-10-01 02:00:04 -0500 (Thu,
01 Oct 2009) | 3 lines Simplify code for porturi, use TRUE/FALSE
constructs when it's just TRUE or FALSE. ................ r221589
| mnicholson | 2009-10-01 10:26:20 -0500 (Thu, 01 Oct 2009) | 9
lines Merged revisions 221588 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r221588 | mnicholson | 2009-10-01 10:24:00 -0500 (Thu, 01 Oct
2009) | 2 lines Use unsigned ints for portinuri flags. ........
................
2009-10-01 16:25 +0000 [r221622] Kevin P. Fleming <kpfleming@digium.com>
* main/udptl.c, /, configs/udptl.conf.sample, UPGRADE.txt: Merged
revisions 221592 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r221592 |
kpfleming | 2009-10-01 11:16:09 -0500 (Thu, 01 Oct 2009) | 12
lines Remove ability to control T.38 FAX error correction from
udptl.conf. chan_sip has had the ability to control T.38 FAX
error correction mode on a per-peer (or global) basis for a
couple of releases now, which is where it should have been all
along. This patch removes the ability to configure it in
udptl.conf, but issues a warning if the user tries to do, telling
them to look at sip.conf.sample for how to configure it now. For
any SIP peers that are T.38 enabled in sip.conf, there is already
a default for FEC error correction even if the user does not
specify any mode, so this change will not turn off error
correction by default, it will have the same default value that
has been in the udptl.conf sample file. ........
2009-09-30 23:07 +0000 [r221477-221485] Matthew Nicholson <mnicholson@digium.com>
* /, channels/chan_sip.c: Merged revisions 221484 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r221484 |
mnicholson | 2009-09-30 18:04:03 -0500 (Wed, 30 Sep 2009) | 2
lines Cleaned up merge from r221432 ........
* configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions
221432 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r221432 | mnicholson | 2009-09-30 15:40:20 -0500 (Wed, 30 Sep
2009) | 17 lines Merged revisions 221360 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r221360 | mnicholson | 2009-09-30 14:36:06 -0500 (Wed, 30 Sep
2009) | 10 lines Fix SRV lookup and Request-URI generation in
chan_sip. This patch adds a new field "portinuri" to the sip
dialog struct and the sip peer struct. That field is used during
RURI generation to determine if the port should be included in
the RURI. It is also used in some places to determine if an SRV
lookup should occur. (closes issue #14418) Reported by: klaus3000
Tested by: klaus3000, mnicholson Review:
https://reviewboard.asterisk.org/r/369/ ........ ................
2009-09-30 21:46 +0000 [r221371-221472] Matthias Nick <mnick@digium.com>
* apps/app_queue.c, /: Merged revisions 221436 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r221436 |
mnick | 2009-09-30 16:15:01 -0500 (Wed, 30 Sep 2009) | 2 lines
Prevents from division by zero ........
* configs/cdr_custom.conf.sample, /, funcs/func_strings.c: Merged
revisions 221368 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r221368 | mnick | 2009-09-30 14:42:36 -0500 (Wed, 30 Sep 2009) |
23 lines Merged revisions 221153,221157,221303 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r221153 | mnick | 2009-09-30 10:37:39 -0500 (Wed, 30 Sep 2009) |
2 lines check bounds - prevents for buffer overflow ........
r221157 | mnick | 2009-09-30 10:41:46 -0500 (Wed, 30 Sep 2009) |
8 lines added a new dialplan function 'CSV_QUOTE' and changed the
cdr_custom.sample.conf (closes issue #15471) Reported by: dkerr
Patches: csv_quote_14.txt uploaded by mnick (license ) Tested by:
mnick ........ r221303 | mnick | 2009-09-30 14:02:00 -0500 (Wed,
30 Sep 2009) | 2 lines changed the prototype definition of
csv_quote ........ ................
2009-09-30 19:15 +0000 [r221304] Terry Wilson <twilson@digium.com>
* configs/sip.conf.sample, main/rtp.c, /, channels/chan_sip.c,
include/asterisk/rtp.h: Merged revisions 221266 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r221266 | twilson | 2009-09-30 12:52:30 -0500 (Wed, 30 Sep 2009)
| 32 lines Merged revisions 221086 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009)
| 25 lines Change the SSRC by default when our media stream
changes Be default, change SSRC when doing an audio stream
changes Asterisk doesn't honor marker bit when reinvited to
already-bridged RTP streams,resulting in far-end stack discarding
packets with "old" timestamps that areactually part of a new
stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is
a reinvite, unless the 'constantssrc' is set to true in sip.conf.
The original issue reported to Digium support detailed the
following situation: ITSP <-> Asterisk 1.4.26.2 <-> SIP-based
Application Server Call comes in fromITSP, Asterisk dials the app
server which sends a re-invite back toAsterisk--not to negotiate
to send media directly to the ITSP, but to indicatethat it's
changing the stream it's sending to Asterisk. The app
servergenerates a new SSRC, sequence numbers, timestamps, and
sets the marker bit on the new stream. Asterisk passes through
the teimstamp of the new stream, butdoes not reset the SSRC,
sequence numbers, or set the marker bit. When the timestamp on
the new stream is older than the timestamp on the originalstream,
the ITSP (which doesn't know there has been any change) discards
the newframes because it thinks they are too old. This patch
addresses this by changing the SSRC on a stream update unless
constantssrc=true is set in sip.conf. Review:
https://reviewboard.asterisk.org/r/374/ ........ ................
2009-09-30 16:57 +0000 [r221204] Tilghman Lesher <tlesher@digium.com>
* main/channel.c, /: Merged revisions 221201 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r221201 | tilghman | 2009-09-30 11:56:42 -0500 (Wed, 30 Sep 2009)
| 14 lines Merged revisions 221200 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r221200 | tilghman | 2009-09-30 11:55:21 -0500 (Wed, 30 Sep 2009)
| 7 lines Avoid a potential NULL dereference. (closes issue
#15865) Reported by: kobaz Patches: 20090915__issue15865.diff.txt
uploaded by tilghman (license 14) Tested by: kobaz ........
................
2009-09-30 14:57 +0000 [r221089] Sean Bright <sean@malleable.com>
* apps/app_voicemail.c, /: Merged revisions 221085 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r221085 | seanbright | 2009-09-30 10:47:58 -0400 (Wed, 30 Sep
2009) | 9 lines Clarify documentation for VoiceMailMain()'s a()
option. We require box numbers, not names as the documentation
implies. (issue #14740) Reported by: pj Patches:
__20090729-app_voicemail-documentation.patch uploaded by lmadsen
(license 10) Tested by: seanbright, lmadsen ........
2009-09-30 04:41 +0000 [r221027-221047] Tilghman Lesher <tlesher@digium.com>
* /, funcs/func_lock.c: Recorded merge of revisions 221044 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r221044 | tilghman | 2009-09-29 23:32:36 -0500 (Tue, 29
Sep 2009) | 8 lines Allow locks to be inherited through a
masquerade without causing starvation. (closes issue #14859)
Reported by: atis Patches: 20090821__issue14859.diff.txt uploaded
by tilghman (license 14) 20090925__issue14859__1.6.1.diff.txt
uploaded by tilghman (license 14) Tested by: atis, tilghman
........
* include/asterisk/smdi.h, include/asterisk/optional_api.h
(removed), apps/app_voicemail.c, include/asterisk/agi.h,
include/asterisk/monitor.h: Remove optional_api from 1.6.2
branch, since it is not currently working. This is a blocking
issue for the 1.6.2 release. (closes issue #15914) Reported by:
mbeckwell Branch:
http://svn.digium.com/svn/asterisk/team/tilghman/optional_api_162
Tested by: mbeckwell
* /, channels/chan_sip.c: Merged revisions 220906 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r220906 | tilghman | 2009-09-29 14:57:37 -0500 (Tue, 29 Sep 2009)
| 16 lines Merged revisions 220873 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r220873 | tilghman | 2009-09-29 12:59:26 -0500 (Tue, 29 Sep 2009)
| 9 lines Reduce CPU usage related to building a peer merely for
devicestates. This fixes a 100% CPU problem in the SIP driver,
found by profiling the driver while the problem was occurring.
(closes issue #14309) Reported by: pkempgen Patches:
20090924__issue14309.diff.txt uploaded by tilghman (license 14)
Tested by: pkempgen, vrban ........ ................
2009-09-29 20:24 +0000 [r220905-220934] Matthew Nicholson <mnicholson@digium.com>
* apps/app_chanspy.c: Avoid a deadlock in chanspy, just in case the
spyee is masqueraded and chanspy_ds_chan_fixup() is called with
the channel locked. (closes issue #15965) Reported by: atis
Patches: chanspy-deadlock-fix1.diff uploaded by mnicholson
(license 96) Tested by: atis
* /, apps/app_confbridge.c: Merged revisions 220904 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r220904 | mnicholson | 2009-09-29 14:49:02 -0500 (Tue, 29 Sep
2009) | 5 lines Fix options 'm' and 's'. They were swapped in the
code. Also document the fact that app_confbridge does not
automatically answer the channel. (closes issue #15964) Reported
by: shrift ........
2009-09-29 17:06 +0000 [r220836] Jeff Peeler <jpeeler@digium.com>
* apps/app_voicemail.c, /: Merged revisions 220833 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r220833 | jpeeler | 2009-09-29 11:58:29 -0500 (Tue, 29 Sep 2009)
| 12 lines Make deletion of temporary greetings work properly
with IMAP_STORAGE When imapgreetings was set to yes, the message
was being deleted but wasn't actually being expunged. When
imapgreetings was set to no, the file based message was not being
deleted at all. All good now! (closes issue #14949) Reported by:
noahisaac Patches: vm_tempgreeting_removal.patch uploaded by
noahisaac (license 748), modified by me ........
2009-09-28 19:13 +0000 [r220725] Sean Bright <sean@malleable.com>
* /, Makefile.rules: Merged revisions 220721 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r220721 | seanbright | 2009-09-28 15:11:20 -0400 (Mon, 28 Sep
2009) | 10 lines Merged revisions 220717 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r220717 | seanbright | 2009-09-28 15:09:25 -0400 (Mon, 28 Sep
2009) | 3 lines When selecting DONT_OPTIMIZE in menuselect,
explicitly pass -O0 to the compiler so we override any default
optimization levels for a particular install. ........
................
2009-09-28 19:11 +0000 [r220722] Jeff Peeler <jpeeler@digium.com>
* /, channels/chan_sip.c: Merged revisions 220718 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r220718 |
jpeeler | 2009-09-28 14:10:10 -0500 (Mon, 28 Sep 2009) | 10 lines
Fix building of registration entry in build_peer when using
callbackextension Check for remotesecret option was
unintentionally always true, which therefore caused the secret
option to never be used. Thanks to dvossel for pointing out the
exact fix. (closes issue #15943) Reported by: tpsast ........
2009-09-27 20:45 +0000 [r220632] Michiel van Baak <michiel@vanbaak.info>
* funcs/func_callerid.c, /: Merged revisions 220629 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r220629 | mvanbaak | 2009-09-27 22:40:16 +0200 (Sun, 27 Sep 2009)
| 3 lines add name argument for the CALLERID dialplan function to
the xml documentation. Pointed out to me on IRC by snuff-home.
Thanks ........
2009-09-26 15:12 +0000 [r220589] Tilghman Lesher <tlesher@digium.com>
* /, include/asterisk/aes.h: Merged revisions 220586 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r220586 | tilghman | 2009-09-26 10:10:28 -0500 (Sat, 26 Sep 2009)
| 2 lines Allow AES to compile, when OpenSSL is not present.
........
2009-09-24 20:38 +0000 [r220369] David Vossel <dvossel@digium.com>
* main/tcptls.c, /: Merged revisions 220365 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r220365 |
dvossel | 2009-09-24 15:37:20 -0500 (Thu, 24 Sep 2009) | 8 lines
fixes tcptls_session memory leak caused by ref count error
(closes issue #15939) Reported by: dvossel Review:
https://reviewboard.asterisk.org/r/375/ ........
2009-09-24 19:42 +0000 [r220292] Tilghman Lesher <tlesher@digium.com>
* apps/app_playback.c, main/pbx.c, /, apps/app_disa.c: Merged
revisions 220289 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r220289 | tilghman | 2009-09-24 14:41:02 -0500 (Thu, 24 Sep 2009)
| 13 lines Merged revisions 220288 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r220288 | tilghman | 2009-09-24 14:39:41 -0500 (Thu, 24 Sep 2009)
| 6 lines Implicitly sending a progress signal breaks some
applications. Call Progress() in your dialplan if you explicitly
want progress to be sent. (Reverts change 216430, closes issue
#15957) Reported by: Pavel Troller on the Asterisk-Dev mailing
list
http://lists.digium.com/pipermail/asterisk-dev/2009-September/039897.html
........ ................
2009-09-24 18:22 +0000 [r220103-220221] Sean Bright <sean@malleable.com>
* Makefile, /: Merged revisions 220217 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r220217 | seanbright | 2009-09-24 14:19:41 -0400 (Thu, 24 Sep
2009) | 9 lines Merged revisions 220213 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r220213 | seanbright | 2009-09-24 14:18:18 -0400 (Thu, 24 Sep
2009) | 1 line Resolve parallel build warnings. Reported by Klaus
Darilion on the asterisk-dev mailing list. ........
................
* Makefile, build_tools/mkpkgconfig, /: Merged revisions 220100 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r220100 | seanbright | 2009-09-24 10:44:08 -0400
(Thu, 24 Sep 2009) | 9 lines Merged revisions 220099 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r220099 | seanbright | 2009-09-24 10:41:57 -0400 (Thu,
24 Sep 2009) | 2 lines Remove the remaining bashisms in the
Makefile/mkpkgconfig ........ ................
2009-09-24 08:43 +0000 [r220031] Michiel van Baak <michiel@vanbaak.info>
* build_tools/mkpkgconfig, /: Merged revisions 220028 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r220028 | mvanbaak | 2009-09-24 10:36:18 +0200
(Thu, 24 Sep 2009) | 14 lines Merged revisions 220027 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r220027 | mvanbaak | 2009-09-24 10:33:50 +0200 (Thu, 24 Sep 2009)
| 7 lines mkpkgconfig does not need bash so make it use /bin/sh
This fixes building on all systems that don't have bash at
/bin/bash Reported by _ys on #asterisk-dev Tested by _ys on
#asterisk-dev ........ ................
2009-09-24 07:45 +0000 [r219989] Tilghman Lesher <tlesher@digium.com>
* apps/app_directory.c, /: Merged revisions 219987 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r219987 | tilghman | 2009-09-24 02:39:44 -0500 (Thu, 24 Sep 2009)
| 8 lines Fix two possible crashes, one only in 1.6.1 and one in
1.6.1 forward. (closes issue #15739) Reported by: DLNoah, jeffg
Patches: 20090914__issue15739.diff.txt uploaded by tilghman
(license 14) 20090922__issue15739.diff.txt uploaded by tilghman
(license 14) Tested by: DLNoah, jeffg ........
2009-09-22 21:48 +0000 [r219821] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c, /: Merged revisions 219818 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r219818 | tilghman | 2009-09-22 16:43:22 -0500
(Tue, 22 Sep 2009) | 17 lines Merged revisions 219816 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r219816 | tilghman | 2009-09-22 16:37:03 -0500 (Tue, 22 Sep 2009)
| 10 lines When IMAP variables were changed during a reload,
Voicemail did not use the new values. This change introduces a
configuration version variable, which ensures that connections
with the old values are not reused but are allowed to expire
normally. (closes issue #15934) Reported by: viniciusfontes
Patches: 20090922__issue15934.diff.txt uploaded by tilghman
(license 14) Tested by: viniciusfontes ........ ................
2009-09-21 17:01 +0000 [r219722] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c, /: Merged revisions 219721 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r219721 | dvossel | 2009-09-21 11:59:05 -0500
(Mon, 21 Sep 2009) | 9 lines Merged revisions 219720 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r219720 | dvossel | 2009-09-21 11:55:53 -0500 (Mon, 21
Sep 2009) | 3 lines Reverting merge 219520. This change was not
necessary. ........ ................
2009-09-20 18:21 +0000 [r219669] Tilghman Lesher <tlesher@digium.com>
* /, main/file.c: Merged revisions 219654 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r219654 | tilghman | 2009-09-20 12:55:49 -0500 (Sun, 20 Sep 2009)
| 15 lines Merged revisions 219653 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r219653 | tilghman | 2009-09-20 12:52:05 -0500 (Sun, 20 Sep 2009)
| 8 lines Really stop the stream, when ast_closestream() is
called. (closes issue #15129) Reported by: bmh Patches:
20090918__issue15129.diff.txt uploaded by tilghman (license 14)
Review: https://reviewboard.asterisk.org/r/372/ ........
................
2009-09-19 03:14 +0000 [r219590] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c, /: Merged revisions 219587 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r219587 | russell | 2009-09-18 21:59:52 -0500
(Fri, 18 Sep 2009) | 13 lines Merged revisions 219586 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r219586 | russell | 2009-09-18 21:51:13 -0500 (Fri, 18 Sep 2009)
| 6 lines Make sure the iax_pvt exists before dereferencing it.
This fixes the latest crash posted on issue 15609. (issue #15609)
........ ................
2009-09-18 23:21 +0000 [r219452-219521] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c, /: Merged revisions 219520 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r219520 | dvossel | 2009-09-18 18:20:58 -0500
(Fri, 18 Sep 2009) | 15 lines Merged revisions 219519 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r219519 | dvossel | 2009-09-18 18:19:50 -0500 (Fri, 18 Sep 2009)
| 9 lines iax2 frame double free The iax frame's retrans sched id
was written over right before iax2_frame_free was called. In
iax2_frame_free that retrans id is used to delete the sched item.
By writing over the retrans field before the sched item could be
deleted, it was possible for a retransmit to occur on a freed
frame. ........ ................
* /, channels/chan_sip.c: Merged revisions 219451 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r219451 | dvossel | 2009-09-18 11:20:41 -0500 (Fri, 18 Sep 2009)
| 20 lines Merged revisions 219450 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r219450 | dvossel | 2009-09-18 11:19:15 -0500 (Fri, 18 Sep 2009)
| 14 lines via-header branches not updated correctly on INVITE
INVITE requests must always contain a new unique branch id. When
a new branch id is created for an INVITE, the dialog's
invite_branch variable must be updated so CANCEL requests use the
correct branch id. (closes issue #15262) Reported by: maniax
Patches: asterisk-1.6.1.0-sip-branch.patch uploaded by tweety
(license 608) invite_new_branch_trunk.diff uploaded by dvossel
(license 671) Tested by: maniax, dvossel ........
................
2009-09-18 13:57 +0000 [r219415] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c, /: Merged revisions 219412 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r219412 | tilghman | 2009-09-18 08:54:51 -0500 (Fri, 18 Sep 2009)
| 6 lines Missing value setting line for maxsecs/maxmessage
(closes issue #15696) Reported by: fhackenberger Patches:
maxsecs.patch uploaded by fhackenberger (license 592) ........
2009-09-17 22:38 +0000 [r219376] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 219371 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r219371 |
dvossel | 2009-09-17 17:37:28 -0500 (Thu, 17 Sep 2009) | 9 lines
fixes deadlock when performing directed pickup w Invite/replaces
(closes issue #15340) Reported by: lmsteffan Patches:
deadlock.patch uploaded by lmsteffan (license 779) Tested by:
lmsteffan ........
2009-09-17 22:37 +0000 [r219370] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 219324 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r219324 | mmichelson | 2009-09-17 17:22:01 -0500 (Thu, 17 Sep
2009) | 12 lines Merged revisions 219320 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r219320 | mmichelson | 2009-09-17 17:20:50 -0500 (Thu, 17 Sep
2009) | 6 lines Send a 100 Trying response when we detect a
spiral. This was problematic during spiral tests at SIPit...
along with some other things as well. ........ ................
2009-09-17 22:06 +0000 [r219307] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 219304 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r219304 | dvossel | 2009-09-17 16:59:21 -0500 (Thu, 17 Sep 2009)
| 27 lines Merged revisions 219303 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r219303 | dvossel | 2009-09-17 16:29:37 -0500 (Thu, 17 Sep 2009)
| 21 lines INVITE w/Replaces deadlock fix This patch cleans up
the locking logic in chan_sip.c's handle_invite_replaces()
function as well as making use of ast_do_masquerade() rather than
forcing the masquerade on an ast_read(). The code had several
redundant unlocks that would result in 'freed more times than
we've locked!' errors. I cleaned these up as well as moving all
the unlock logic to the end of the function. This patch should
also resolve the issue people were having with the replacecall
channel never being unlocked with one legged calls. (closes issue
#15151) Reported by: irroot Patches: invite_w_replaces_1.4.diff
uploaded by dvossel (license 671) Tested by: irroot, dvossel
Review: https://reviewboard.asterisk.org/r/371/ ........
................
2009-09-17 19:58 +0000 [r219267] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 219264 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r219264 |
file | 2009-09-17 14:57:39 -0500 (Thu, 17 Sep 2009) | 2 lines
Ensure no spaces exist before "refresher=" when doing the
comparison. ........
2009-09-17 Leif Madsen <lmadsen@digium.com>
* Released Asterisk 1.6.2.0-rc2
2009-09-17 15:38 +0000 [r219194] Matthew Nicholson <mnicholson@digium.com>
* main/channel.c, /, include/asterisk/cdr.h,
include/asterisk/channel.h: Merged revisions 219139 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r219139 | mnicholson | 2009-09-17 10:18:01 -0500
(Thu, 17 Sep 2009) | 17 lines Merged revisions 219136 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r219136 | mnicholson | 2009-09-17 09:58:39 -0500 (Thu, 17 Sep
2009) | 10 lines Prevent a potential race condition and crash
when hanging up a channel by removing the channel from the
channel list before begining channel tear down. This fix may
potentially cause problems with CDR backends that access the
channel a CDR is associated with via the channel list. This fix
makes the channel unavabile at the time when the CDR backend is
invoked. This has been documented in include/asterisk/cdr.h.
(closes issue #15316) Reported by: vmarrone Tested by: mnicholson
Review: https://reviewboard.asterisk.org/r/362/ ........
................
2009-09-16 23:52 +0000 [r219063] Tilghman Lesher <tlesher@digium.com>
* main/config.c, configs/extensions.conf.sample, /: Merged
revisions 219061 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r219061 | tilghman | 2009-09-16 18:42:12 -0500 (Wed, 16 Sep 2009)
| 15 lines Merged revisions 219023 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r219023 | tilghman | 2009-09-16 18:21:53 -0500 (Wed, 16 Sep 2009)
| 8 lines Properly deal with quotes in the arguments of '#exec'
includes. (closes issue #15583) Reported by: pkempgen Patches:
20090726__issue15583.diff.txt uploaded by tilghman (license 14)
20090726__issue15583-1.4-4.diff.txt uploaded by pkempgen (license
169) Tested by: pkempgen ........ ................
2009-09-16 19:40 +0000 [r218938] David Brooks <dbrooks@digium.com>
* main/pbx.c, /: Merged revisions 218868 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r218868 | dbrooks | 2009-09-16 13:06:42 -0500 (Wed, 16 Sep 2009)
| 20 lines Merged revisions 218867 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r218867 | dbrooks | 2009-09-16 13:00:45 -0500 (Wed, 16 Sep 2009)
| 13 lines Fixes CID pattern matching behavior to mirror that of
extension pattern matching. Pattern matching for extensions uses
a type of scoring system, giving values for specificity to each
character in the pattern. Unfortunately, this is done character
by character, in order. This does lead to some less specific
patterns being first in line for matching, but it will usually
get the job done. This patch merely brings CID matching to the
same level as extension matching. This patch does not attempt to
tackle the problem shared by extension matching. (closes issue
#14708) Reported by: klaus3000 ........ ................
2009-09-16 19:29 +0000 [r218937] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 218933 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r218933 |
mmichelson | 2009-09-16 14:25:36 -0500 (Wed, 16 Sep 2009) | 12
lines Reverse order of args to fread. This way, we don't always
write a null byte into byte 1 of the buffer (closes issue #15905)
Reported by: ebroad Patches: freadfix.patch uploaded by ebroad
(license 878) Tested by: ebroad ........
2009-09-16 19:25 +0000 [r218934] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 218918 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r218918 |
file | 2009-09-16 13:31:47 -0500 (Wed, 16 Sep 2009) | 5 lines On
TCP and TLS connections do not attempt to stop retransmission of
the packet internally. This was preventing responses from being
properly processed because the packet was not being found causing
handle_response to return prematurely. ........
2009-09-16 13:38 +0000 [r218802] Russell Bryant <russell@digium.com>
* contrib/firmware/iax/iaxy.bin (removed), /, UPGRADE.txt: Merged
revisions 218799 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r218799 | russell | 2009-09-16 08:34:41 -0500 (Wed, 16 Sep 2009)
| 16 lines Merged revisions 218798 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r218798 | russell | 2009-09-16 08:33:43 -0500 (Wed, 16 Sep 2009)
| 9 lines Remove the IAXy firmware from Asterisk. The firmware
can now be found on downloads.digium.com, where the rest of our
binary downloads live. This was the last part of our Asterisk
tarballs that was considered non-free by Debian. :-) (closes
issue #15838) Reported by: paravoid ........ ................
2009-09-15 22:46 +0000 [r218733] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c, /: Merged revisions 218731 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r218731 | tilghman | 2009-09-15 17:33:10 -0500
(Tue, 15 Sep 2009) | 13 lines Merged revisions 218730 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r218730 | tilghman | 2009-09-15 17:27:41 -0500 (Tue, 15 Sep 2009)
| 6 lines If the user enters the same password as before, don't
signal an error when the change does nothing. (closes issue
#15492) Reported by: cbbs70a Patches:
20090713__issue15492.diff.txt uploaded by tilghman (license 14)
........ ................
2009-09-15 19:24 +0000 [r218688] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 218687 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r218687 |
dvossel | 2009-09-15 14:22:37 -0500 (Tue, 15 Sep 2009) | 2 lines
upward bound checking for port string to int conversion ........
2009-09-15 16:18 +0000 [r218590] Matthew Nicholson <mnicholson@digium.com>
* /, channels/chan_sip.c: Merged revisions 218586 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r218586 | mnicholson | 2009-09-15 11:15:02 -0500 (Tue, 15 Sep
2009) | 15 lines Merged revisions 218578 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r218578 | mnicholson | 2009-09-15 11:03:54 -0500 (Tue, 15 Sep
2009) | 8 lines Send request contact header field with response
to registrer queries instead of the address of record. (closes
issue #14438) Reported by: ravindrad Patches: regquerypatch
uploaded by ravindrad (license 684) Tested by: ravindrad ........
................
2009-09-15 16:06 +0000 [r218582] Tilghman Lesher <tlesher@digium.com>
* apps/app_followme.c, /: Merged revisions 218579 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r218579 | tilghman | 2009-09-15 11:04:41 -0500 (Tue, 15 Sep 2009)
| 16 lines Merged revisions 218577 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r218577 | tilghman | 2009-09-15 11:01:17 -0500 (Tue, 15 Sep 2009)
| 9 lines Ensure FollowMe sets language in channels it creates.
Also, not in the original bug report, but related fields are
accountcode and musicclass, and the inheritance of datastores.
(closes issue #15372) Reported by: Romik Patches:
20090828__issue15372.diff.txt uploaded by tilghman (license 14)
Tested by: cervajs ........ ................
2009-09-15 15:59 +0000 [r218576] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 218430 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r218430 | jpeeler | 2009-09-14 17:38:25 -0500
(Mon, 14 Sep 2009) | 18 lines Merged revisions 218401 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r218401 | jpeeler | 2009-09-14 16:47:11 -0500 (Mon, 14 Sep 2009)
| 11 lines Fix handling of DAHDI_EVENT_REMOVED event to prevent
crash in do_monitor. After talking to rmudgett about some of his
recent iflist locking changes, it was determined that the only
place that would destroy a channel without being explicitly to do
so was in handle_init_event. The loop to walk the interface list
has been modified to wait to destroy the channel until the
dahdi_pvt of the channel to be destroyed is no longer needed.
(closes issue #15378) Reported by: samy ........ ................
2009-09-15 15:42 +0000 [r218507-218575] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 218566 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r218566 |
mmichelson | 2009-09-15 10:40:14 -0500 (Tue, 15 Sep 2009) | 4
lines Use a better method of ensuring null-termination of the
buffer while reading the SDP when using TCP. ........
* /, channels/chan_sip.c: Merged revisions 218499,218504 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r218499 | mmichelson | 2009-09-15 09:59:50 -0500 (Tue,
15 Sep 2009) | 3 lines Fix off-by-one error when reading SDP sent
over TCP. ........ r218504 | mmichelson | 2009-09-15 10:05:53
-0500 (Tue, 15 Sep 2009) | 3 lines Ensure that SDP read from TCP
socket is null-terminated. ........
2009-09-15 15:05 +0000 [r218503] Kevin P. Fleming <kpfleming@digium.com>
* sounds/Makefile, /: Merged revisions 218500 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r218500 | kpfleming | 2009-09-15 11:02:21 -0400 (Tue, 15 Sep
2009) | 9 lines Merged revisions 218497 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r218497 | kpfleming | 2009-09-15 10:55:58 -0400 (Tue, 15 Sep
2009) | 1 line Use proper hostname for downloading sound files.
........ ................
2009-09-14 19:49 +0000 [r218364] Tilghman Lesher <tlesher@digium.com>
* sounds/Makefile, apps/app_voicemail.c, /,
configs/voicemail.conf.sample: Merged revisions 218361 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r218361 | tilghman | 2009-09-14 14:29:48 -0500
(Mon, 14 Sep 2009) | 11 lines Recorded merge of revisions 218331
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r218331 | tilghman | 2009-09-14 14:16:35 -0500 (Mon, 14 Sep 2009)
| 4 lines Don't say "Please try again" if we don't give the user
another chance to try again. (issue #15055, SWP-129) Reported by:
jthurman ........ ................
2009-09-14 18:18 +0000 [r218300] Joshua Colp <jcolp@digium.com>
* /, main/features.c: Merged revisions 218295 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r218295 |
file | 2009-09-14 13:16:39 -0500 (Mon, 14 Sep 2009) | 2 lines Do
not attempt to add a parking extension if an error occurred while
reading the configuration. ........
2009-09-14 15:20 +0000 [r218238] Matthew Nicholson <mnicholson@digium.com>
* /, apps/app_directed_pickup.c: Merged revisions 218224 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r218224 | mnicholson | 2009-09-14 09:57:23 -0500
(Mon, 14 Sep 2009) | 14 lines Merged revisions 218223 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r218223 | mnicholson | 2009-09-14 09:53:57 -0500 (Mon, 14 Sep
2009) | 8 lines Ensure we don't pickup ourselves when doing
pickup by exten. (closes issue #15100) Reported by: lmsteffan
Patches: (modified) pickup.patch uploaded by lmsteffan (license
779) ........ ................
2009-09-13 22:12 +0000 [r218219] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* channels/chan_phone.c, /: gcc 4.4: Remove a nop memset size 0
that annoys gcc This memset doesn't write beyond the end of the
buffer. (tmpbuf has size of 4). Merged revisions 218184 via
svnmerge from http://svn.digium.com/svn/asterisk/trunk
2009-09-13 05:59 +0000 [r218151] Moises Silva <moises.silva@gmail.com>
* channels/chan_dahdi.c, /: Merged revisions 218150 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r218150 | moy | 2009-09-13 01:51:46 -0400 (Sun, 13 Sep 2009) | 1
line get rid of mfcr2 monitor thread condition, is problematic
........
2009-09-11 06:00 +0000 [r217926-218055] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c, /: Merged revisions 218050 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r218050 |
tilghman | 2009-09-11 00:58:11 -0500 (Fri, 11 Sep 2009) | 3 lines
Check the origination priority for more matches, not the current
priority. Found by Pavel Troller on the -dev list. ........
* apps/app_queue.c, /: Merged revisions 217990 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r217990 | tilghman | 2009-09-10 18:54:51 -0500 (Thu, 10 Sep 2009)
| 10 lines Merged revisions 217989 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r217989 | tilghman | 2009-09-10 18:52:22 -0500 (Thu, 10 Sep 2009)
| 3 lines Don't ring another channel, if there's not enough time
for a queue member to answer. (Fixes AST-228) ........
................
* channels/chan_iax2.c, contrib/scripts/iax-friends.sql, /,
channels/chan_sip.c: Merged revisions 217916 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r217916 |
tilghman | 2009-09-10 18:12:16 -0500 (Thu, 10 Sep 2009) | 2 lines
Make calltoken support work with realtime users and peers.
........
2009-09-10 21:21 +0000 [r217821] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c, /: Merged revisions 217807 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r217807 | dvossel | 2009-09-10 16:07:47 -0500
(Thu, 10 Sep 2009) | 28 lines Merged revisions 217806 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r217806 | dvossel | 2009-09-10 16:06:07 -0500 (Thu, 10 Sep 2009)
| 22 lines IAX2 encryption regression The IAX2 Call Token
security patch inadvertently broke the use of encryption due to
the reorganization of code in the socket_process() function. When
encryption is used, an incoming full frame must first be
decrypted before the information elements can be parsed. The
security release mistakenly moved IE parsing before decryption in
order to process the new Call Token IE. To resolve this,
decryption of full frames is once again done before looking into
the frame. This involves searching for an existing callno,
checking the pvt to see if encryption is turned on, and
decrypting the packet before the internal fields of the full
frame are accessed. (closes issue #15834) Reported by: karesmakro
Patches: iax2_encryption_fix_1.4.diff uploaded by dvossel
(license 671) Tested by: dvossel, karesmakro Review:
https://reviewboard.asterisk.org/r/355/ ........ ................
2009-09-10 19:56 +0000 [r217739] mnick <mnick@localhost>:
* res/res_musiconhold.c, /: Merged revisions 217730 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r217730 | mnick | 2009-09-10 14:39:41 -0500 (Thu, 10 Sep 2009) |
17 lines Sets the correct musicclass after an announcement
(closes issue #15279) Reported by: mbeckwell Patches: patch.txt
uploaded by mnick (license ) Tested by: mnick (closes issue
#15832) Reported by: mbeckwell Patches: patch.txt uploaded by
mnick (license 874) Tested by: mnick ........
2009-09-10 18:40 +0000 [r217665] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Merged revisions 216805 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r216805 |
oej | 2009-09-07 18:08:08 +0200 (MÃ¥n, 07 Sep 2009) | 2 lines
Since it's possible to have more than 999 calls, I'm changing the
call counter roof to something higher. ........
2009-09-10 18:19 +0000 [r217647] Tilghman Lesher <tlesher@digium.com>
* res/res_config_odbc.c, /, configure,
include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
217638 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r217638 |
tilghman | 2009-09-10 13:17:14 -0500 (Thu, 10 Sep 2009) | 4 lines
Verify support for wide ODBC character types before using them.
(closes issue #15870) Reported by: nic_bellamy ........
2009-09-10 15:14 +0000 [r217632] Moises Silva <moises.silva@gmail.com>
* channels/chan_dahdi.c, /: Merged revisions 217524 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r217524 | moy | 2009-09-09 17:48:04 -0400 (Wed, 09 Sep 2009) | 1
line ast_log replaced for ast_verbose in MFCR2 event
notifications ........
2009-09-10 12:09 +0000 [r217594] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Merged revisions 217593 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r217593 |
oej | 2009-09-10 14:06:55 +0200 (Tor, 10 Sep 2009) | 8 lines
Include ActionID in all events that are responsed to AMI Action
SIPShowRegistry (closes issue #15868) Reported by: nic_bellamy
Patches: manager_SIPshowregistry_actionid.patch uploaded by nic
bellamy (license 299) ........
2009-09-09 20:37 +0000 [r217519] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* /, res/res_phoneprov.c: gcc 4.4 fix: union instead of cast gcc
4.4 has more strict rules for aliasing. It doesn't like a struct
sockaddr_in pointer pointing to a struct sockaddr. So we make it
a union. Merged revisions 217445 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
2009-09-09 10:58 +0000 [r217369] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Merged revisions 217368 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r217368 |
oej | 2009-09-09 12:39:43 +0200 (Ons, 09 Sep 2009) | 2 lines Not
having any TLS session to write to is a serious XMIT_ERROR.
........
2009-09-08 22:20 +0000 [r217299] Sean Bright <sean@malleable.com>
* /, apps/app_meetme.c: Merged revisions 217286 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r217286 |
seanbright | 2009-09-08 18:17:08 -0400 (Tue, 08 Sep 2009) | 4
lines Fix compilation of app_meetme. Reported by ebroad in
#asterisk-bugs ........
2009-09-08 20:33 +0000 [r217217] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_meetme.c: Merged revisions 217199 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r217199 | tilghman | 2009-09-08 15:28:41 -0500 (Tue, 08 Sep 2009)
| 14 lines Merged revisions 217156 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r217156 | tilghman | 2009-09-08 15:01:45 -0500 (Tue, 08 Sep 2009)
| 7 lines When MOH is playing on the channel, announcements sent
through the conference are not heard. (closes issue #14588)
Reported by: voipas Patches: 20090716__issue14588__2.diff.txt
uploaded by tilghman (license 14) Tested by: lmadsen, twisted,
tilghman ........ ................
2009-09-08 16:39 +0000 [r217077] Kevin P. Fleming <kpfleming@digium.com>
* /, configure, include/asterisk/autoconfig.h.in, configure.ac:
Merged revisions 217074 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r217074 |
kpfleming | 2009-09-08 11:37:28 -0500 (Tue, 08 Sep 2009) | 9
lines Ensure that the default autoconf CFLAGS are not used. A
recent change to the configure script that allows the user to
specify CFLAGS and/or LDFLAGS to the script had the unfortunate
side effect of letting autoconf's default CFLAGS (-g -O2) feed in
to the rest of the build system, thereby overriding the
DONT_OPTIMIZE setting in menuselect. That problem is now
corrected. ........
2009-09-08 15:36 +0000 [r217036] Tilghman Lesher <tlesher@digium.com>
* /, res/res_limit.c: Merged revisions 217033 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r217033 |
tilghman | 2009-09-08 10:30:18 -0500 (Tue, 08 Sep 2009) | 4 lines
Remove what appears to be an unnecessary define. (closes issue
#15851) Reported by: tzafrir ........
2009-09-08 14:27 +0000 [r216994] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 216993 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r216993 |
dvossel | 2009-09-08 09:26:30 -0500 (Tue, 08 Sep 2009) | 14 lines
caller id number empty parse_uri was not being given the correct
scheme's, as a result, uri parsing did not parse the username
correctly. One of the side effects of this is an empty caller id.
(closes issue #15839) Reported by: ebroad Patches:
blank_cidv2.patch uploaded by ebroad (license 878)
parse_uri_fix.diff uploaded by dvossel (license 671) Tested by:
ebroad, dvossel ........
2009-09-07 16:43 +0000 [r216647-216845] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Merged revisions 216842 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r216842 |
oej | 2009-09-07 18:35:12 +0200 (MÃ¥n, 07 Sep 2009) | 2 lines
Make sure we reset global_exclude_static at channel reload
........
* /, channels/chan_sip.c: Merged revisions 216695 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r216695 |
oej | 2009-09-07 15:06:19 +0200 (MÃ¥n, 07 Sep 2009) | 8 lines If
there is no session timer in the INVITE, set it to default value
(not unset minimum = -1) Patch by oej closes issue #15621
Reported by: fnordian Tested by: atis ........
* CHANGES, UPGRADE.txt: Add docs
* configs/sip.conf.sample, apps/app_playback.c, main/pbx.c, /,
channels/chan_sip.c, apps/app_disa.c: Merged revisions 216438 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r216438 | oej | 2009-09-04 16:02:34 +0200 (Fre,
04 Sep 2009) | 35 lines Merged revisions 216430 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27
lines Make apps send PROGRESS control frame for early media and
fix too early media issue in SIP The issue at hand is that some
legacy (dying) PBX systems send empty media frames on PRI links
*before* any call progress. The SIP channel receives these frames
and by default signals 183 Session progress and starts sending
media. This will cause phones to play silence and ignore the
later 180 ringing message. A bad user experience. The fix is
twofold: - We discovered that asterisk apps that support early
media ("noanswer") did not send any PROGRESS frame to indicate
early media. Fixed. - We introduce a setting in chan_sip so that
users can disable any relay of media frames before the outbound
channel actually indicates any sort of call progress. In 1.4,
1.6.0 and 1.6.1, this will be disabled for backward
compatibility. In later versions of Asterisk, this will be
enabled. We don't assume that it will change your Asterisk phone
experience - only for the better. We encourage third-party
application developers to make sure that if they have
applications that wants to send early media, add a PROGRESS
control frame transmission to make sure that all channel drivers
actually will start sending early media. This has not been the
default in Asterisk previous to this patch, so if you got
inspiration from our code, you need to update accordingly. Sorry
for the trouble and thanks for your support. This code has been
running for a few months in a large scale installation (over 250
servers with PRI and/or BRI links to old PBX systems). That's no
proof that this is an excellent patch, but, well, it's tested :-)
........ ................
2009-09-04 19:42 +0000 [r216598] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 216594 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r216594 |
dvossel | 2009-09-04 14:32:07 -0500 (Fri, 04 Sep 2009) | 7 lines
sip peer matching by address only with TCP/TLS This patch removes
the contact header matching logic and adds logic to match all
tcp/tls connections by ip only Review:
https://reviewboard.asterisk.org/r/354/ ........
2009-09-04 19:32 +0000 [r216597] Sean Bright <sean@malleable.com>
* apps/app_voicemail.c, /: Merged revisions 216593 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r216593 | seanbright | 2009-09-04 15:29:02 -0400 (Fri, 04 Sep
2009) | 1 line Use ast_free() instead of free(). ........
2009-09-04 17:53 +0000 [r216550-216553] Tilghman Lesher <tlesher@digium.com>
* /, include/asterisk/lock.h: Merged revisions 216551 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r216551 | tilghman | 2009-09-04 12:50:21 -0500 (Fri, 04 Sep 2009)
| 2 lines Fix trunk breakage. ........
* UPGRADE-1.6.txt, main/pbx.c, /: Merged revisions 216547 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r216547 | tilghman | 2009-09-04 12:31:44 -0500 (Fri, 04
Sep 2009) | 3 lines Enable turning off the application delimiter
warning with the 'dontwarn' option. Suggested on the -dev list,
and implemented in an alternate way by me. ........
2009-09-04 15:11 +0000 [r216469-216509] Michiel van Baak <michiel@vanbaak.info>
* /, main/utils.c: Merged revisions 216506 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r216506 | mvanbaak | 2009-09-04 17:05:05 +0200 (Fri, 04 Sep 2009)
| 9 lines Merged revisions 216435 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r216435 | mvanbaak | 2009-09-04 15:56:10 +0200 (Fri, 04 Sep 2009)
| 2 lines make asterisk compile under devmode with DEBUG_THREADS
enabled on OpenBSD ........ ................
* /, include/asterisk/lock.h: Merged revisions 216437 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r216437 | mvanbaak | 2009-09-04 16:00:38 +0200 (Fri, 04 Sep 2009)
| 2 lines make sure canlog is set so we can compile with
DEBUG_THREADS enabled on OpenBSD ........
2009-09-04 13:57 +0000 [r216267-216436] Russell Bryant <russell@digium.com>
* /, channels/chan_sip.c: Merged revisions 216368 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r216368 |
russell | 2009-09-04 08:14:25 -0500 (Fri, 04 Sep 2009) | 12 lines
Do not treat every SIP peer as if they were configured with
insecure=port. There was a problem in the function responsible
for doing peer matching by IP address and port number such that
during the second pass for checking for a peer configured with
insecure=port, it would end up treating every peer as if it had
been configured that way. These changes fix the logic in the peer
IP and port comparison callback to handle insecure=port checking
properly. This problem was introduced when SIP peers were
converted to astobj2. Many thanks to dvossel for noticing this
while working on another peer matching issue. ........
* doc/IAX2-security.txt (added), /: Merged revisions 216264 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r216264 | russell | 2009-09-04 05:48:44 -0500
(Fri, 04 Sep 2009) | 16 lines Merged revisions 216263 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
................ r216263 | russell | 2009-09-04 05:48:00 -0500
(Fri, 04 Sep 2009) | 9 lines Merged revisions 216262 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r216262 | russell | 2009-09-04 05:47:37 -0500 (Fri, 04
Sep 2009) | 2 lines Add a plain text version of the IAX2 security
document. ........ ................ ................
2009-09-04 06:14 +0000 [r216225] Michiel van Baak <michiel@vanbaak.info>
* main/astobj2.c, /: Merged revisions 216222 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r216222 |
mvanbaak | 2009-09-04 08:08:33 +0200 (Fri, 04 Sep 2009) | 3 lines
make sure 'start' is always initialized. Makes asterisk compile
with --enable-dev-mode ........
2009-09-03 19:44 +0000 [r216014-216099] Russell Bryant <russell@digium.com>
* /, UPGRADE.txt: Merged revisions 216092 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r216092 | russell | 2009-09-03 14:38:35 -0500 (Thu, 03 Sep 2009)
| 16 lines Merged revisions 216085 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
................ r216085 | russell | 2009-09-03 14:36:46 -0500
(Thu, 03 Sep 2009) | 9 lines Merged revisions 216080 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r216080 | russell | 2009-09-03 14:35:23 -0500 (Thu, 03
Sep 2009) | 2 lines Add a note about IAX2 to UPGRADE.txt.
........ ................ ................
* /, doc/IAX2-security.pdf (added): Merged revisions 216009 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r216009 | russell | 2009-09-03 13:45:54 -0500
(Thu, 03 Sep 2009) | 16 lines Merged revisions 216008 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
................ r216008 | russell | 2009-09-03 13:44:58 -0500
(Thu, 03 Sep 2009) | 9 lines Merged revisions 216005 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r216005 | russell | 2009-09-03 13:42:24 -0500 (Thu, 03
Sep 2009) | 2 lines Add IAX2 security document related to
AST-2009-006. ........ ................ ................
2009-09-03 18:42 +0000 [r216007] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c, channels/iax2-parser.c, main/astobj2.c,
configs/iax.conf.sample, include/asterisk/acl.h,
channels/iax2-parser.h, /, include/asterisk/astobj2.h,
channels/iax2.h, main/acl.c: Merged revisions 215955 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r215955 | dvossel | 2009-09-03 11:31:54 -0500 (Thu, 03 Sep 2009)
| 6 lines Merge code associated with AST-2009-006 (closes issue
#12912) Reported by: rathaus Tested by: tilghman, russell,
dvossel, dbrooks ........
2009-09-03 14:21 +0000 [r215887-215929] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Merged revisions 215891 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r215891 |
oej | 2009-09-03 15:02:41 +0200 (Tor, 03 Sep 2009) | 10 lines Add
known internal IP address when autodomain=yes (closes issue
#14573) Reported by: pj Patches: sip-internip-autodomain1.diff
uploaded by mnicholson (license 96) modified by oej Tested by: pj
........
* main/rtp.c, channels/chan_sip.c: Fix bad reports in "sip show
channelstats". Not directly mergeable in svn trunk, needs more
tests, therefore committed directly to 1.6.2. (closes issue
#15819) Reported by: klaus3000 Patches:
asterisk-1.6.2-beta4-sipshowchannelstats-patch-0.2.txt uploaded
by klaus3000 (license 65) Tested by: klaus3000, oej
2009-09-03 06:02 +0000 [r215841] Michiel van Baak <michiel@vanbaak.info>
* doc/manager_1_1.txt, /: Merged revisions 215838 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r215838 |
mvanbaak | 2009-09-03 07:57:23 +0200 (Thu, 03 Sep 2009) | 5 lines
Document that SIPshowpeer and SKINNYshowline now include the
configured parkinglot in their response. Prodded by snuff-work on
#asterisk-dev IRC channel ........
2009-09-03 03:44 +0000 [r215802] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_sip.c: Merged revisions 215801 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r215801 |
tilghman | 2009-09-02 22:43:51 -0500 (Wed, 02 Sep 2009) | 5 lines
Default the callback extension to "s". This is a regression.
(closes issue #15764) Reported by: elguero Change-type: bugfix
........
2009-09-03 00:34 +0000 [r215795] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c: Merged revisions 215758 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r215758 | twilson | 2009-09-02 18:31:04 -0500 (Wed, 02 Sep 2009)
| 25 lines Merged revisions 215682 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r215682 | twilson | 2009-09-02 16:41:22 -0500 (Wed, 02 Sep 2009)
| 18 lines Re-send non-100 provisional responses to prevent
cancellation From section 13.3.1.1 of RFC 3261: If the UAS
desires an extended period of time to answer the INVITE, it will
need to ask for an "extension" in order to prevent proxies from
canceling the transaction. A proxy has the option of canceling a
transaction when there is a gap of 3 minutes between responses in
a transaction. To prevent cancellation, the UAS MUST send a
non-100 provisional response at every minute, to handle the
possibility of lost provisional responses. (closes issue #11157)
Reported by: rjain Tested by: twilson Review:
https://reviewboard.asterisk.org/r/315/ ........ ................
2009-09-02 21:46 +0000 [r215683] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 215681 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r215681 |
dvossel | 2009-09-02 16:39:31 -0500 (Wed, 02 Sep 2009) | 10 lines
port string to int conversion using sscanf There are several
instances where a port is parsed from a uri or some other source
and converted to an int value using atoi(), if for some reason
the port string is empty, then a standard port is used. This
logic is used over and over, so I created a function to handle it
in a safer way using sscanf(). ........
2009-09-02 21:37 +0000 [r215647-215680] Michiel van Baak <michiel@vanbaak.info>
* /, channels/chan_sip.c, channels/chan_skinny.c: Merged revisions
215665 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r215665 |
mvanbaak | 2009-09-02 23:23:17 +0200 (Wed, 02 Sep 2009) | 5 lines
add Parkinglot info to sip show peer <foo> and skinny show line
<foo> If we had this from the start, debugging the 'parking not
using configured parkinglot' bug would have been easier. ........
* /, main/features.c: Merged revisions 215622 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r215622 |
mvanbaak | 2009-09-02 22:21:51 +0200 (Wed, 02 Sep 2009) | 4 lines
- lock channel before looking for a channel variable - Init the
parkings list member of struct parkinglot. Thanks Sean for the
explanation why this should be here. ........
2009-09-02 18:52 +0000 [r215569-215570] Tilghman Lesher <tlesher@digium.com>
* /, main/Makefile, main/app.c: Merged revisions 215567 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r215567 | tilghman | 2009-09-02 13:37:25 -0500 (Wed, 02
Sep 2009) | 9 lines Close up to the soft open file limit (same on
Linux, but varies drastically on OS X). Also, a Makefile fix for
Darwin (OS X). (closes issue #14542) Reported by: jtodd Patches:
20090901__issue14542.diff.txt uploaded by tilghman (license 14)
Tested by: jtodd, tilghman Change-type: bugfix ........
* /, channels/chan_sip.c: Merged revisions 215222 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r215222 |
tilghman | 2009-09-01 16:19:40 -0500 (Tue, 01 Sep 2009) | 3 lines
Fix register such that lines with a transport string, but without
an authuser, parse correctly. (AST-228) ........
2009-09-02 17:35 +0000 [r215523] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 215522 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r215522 |
dvossel | 2009-09-02 12:26:40 -0500 (Wed, 02 Sep 2009) | 11 lines
SIP uri parsing cleanup Now, the scheme passed to parse_uri can
either be a single scheme, or a list of schemes ',' delimited.
This gets rid of the whole problem of having to create two
buffers and calling parse_uri twice to check for separate
schemes. Review: https://reviewboard.asterisk.org/r/343/ ........
2009-09-02 16:35 +0000 [r215512] Michiel van Baak <michiel@vanbaak.info>
* /, channels/chan_skinny.c: Merged revisions 215479 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r215479 | mvanbaak | 2009-09-02 18:20:23 +0200 (Wed, 02 Sep 2009)
| 3 lines like in chan_sip's sip_new skinny should copy the
configured parkinglot from a line to the newly created channel.
This makes callparking honor the configured parkinglot for skinny
lines as well. ........
2009-09-02 16:09 +0000 [r215467] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 215466 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r215466 |
dvossel | 2009-09-02 11:08:00 -0500 (Wed, 02 Sep 2009) | 16 lines
SIP support for keep-alive event keep-alive events are used by
Sipura/Linksys for NAT keepalive. There currently don't appear to
be any problems with NAT, but everytime a keep-alive event is
received, Asterisk responds with a "489 Bad event". This error
may indicate to a user that NAT problems exist just because this
even is not supported. Now, rather than respond with an error,
the packet is consumed and a "200 ok" is sent just to indicate we
received the packet. (issue #15084) Patches:
chan_sip.keepalive.v1.diff uploaded by IgorG (license 20)
........
2009-09-02 16:07 +0000 [r215465] Michiel van Baak <michiel@vanbaak.info>
* /, channels/chan_sip.c: Merged revisions 215462 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r215462 |
mvanbaak | 2009-09-02 17:56:46 +0200 (Wed, 02 Sep 2009) | 12
lines Honor configured parkinglot when parking and retrieving
parked calls Thank oej for pointing out the fact that sip_new did
not copy parkinglot from the peer into the newly created channel.
(closes issue #15538) Reported by: gracedman Patches:
2009090100_sipnewparkinglot-161.diff.txt uploaded by mvanbaak
(license 7) With mod by me to also fix callparking as well (this
uploaded patch only fixed retrieving a parked call) Tested by:
gracedman, mvanbaak ........
2009-09-02 01:49 +0000 [r215376] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
* /, apps/app_softhangup.c: Merged revisions 215338 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r215338 | dhubbard | 2009-09-01 20:16:59 -0500
(Tue, 01 Sep 2009) | 18 lines Merged revisions 215270 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r215270 | dhubbard | 2009-09-01 18:04:52 -0500 (Tue, 01 Sep 2009)
| 12 lines Use strrchr() so SoftHangup will correctly truncate
multi-hyphen channel names In general channel names are in the
form Foo/Bar-Z, but the channel name could have multiple hyphens
and look like Foo/B-a-r-Z. Use strrchr to truncate the channel
name at the last hyphen. (closes issue #15810) Reported by:
dhubbard Patches: dw-softhangup-1.4.patch uploaded by dhubbard
(license 733) ........ ................
2009-09-01 20:00 +0000 [r215165] Kevin P. Fleming <kpfleming@digium.com>
* main/frame.c, /: Merged revisions 215161 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r215161 |
kpfleming | 2009-09-01 14:50:48 -0500 (Tue, 01 Sep 2009) | 3
lines Ensure that frame dumps of AST_CONTROL_T38_PARAMETERS
frames are properly decoded. ........
2009-08-31 16:22 +0000 [r214822-214960] Tilghman Lesher <tlesher@digium.com>
* channels/chan_local.c, /: Merged revisions 214945 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r214945 | tilghman | 2009-08-31 11:18:33 -0500
(Mon, 31 Aug 2009) | 14 lines Merged revisions 214940 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r214940 | tilghman | 2009-08-31 11:16:52 -0500 (Mon, 31 Aug 2009)
| 7 lines Also unlock the "other" channel, when returning, due to
glare. (closes issue #15787) Reported by: tim_ringenbach Patches:
chan_local.diff uploaded by tim ringenbach (license 540) Tested
by: tim_ringenbach ........ ................
* Makefile, /: Merged revisions 214898 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r214898 |
tilghman | 2009-08-30 17:10:35 -0500 (Sun, 30 Aug 2009) | 2 lines
Force Darwin on ppc platforms to compile with a target level that
supports aliasing. ........
* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
pbx/pbx_lua.c: Merged revisions 214819 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r214819 |
tilghman | 2009-08-30 01:43:04 -0500 (Sun, 30 Aug 2009) | 4 lines
If lua is detected with the lua5.1 prefix (or not), adjust the
include path accordingly. Based upon feedback to a release
announcement on the -users list. See
http://lists.digium.com/pipermail/asterisk-users/2009-August/236954.html
........
2009-08-29 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.6.2.0-rc1 released.
2009-08-28 20:17 +0000 [r214707] Tilghman Lesher <tlesher@digium.com>
* main/channel.c, /: Merged revisions 214702 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r214702 | tilghman | 2009-08-28 15:14:39 -0500 (Fri, 28 Aug 2009)
| 15 lines Merged revisions 214701 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r214701 | tilghman | 2009-08-28 15:13:32 -0500 (Fri, 28 Aug 2009)
| 8 lines Modify comment to be a bit more accurate. We have kept
this comment around long enough, that it's pretty clear that
we're keeping the code, because changing the code would require a
pretty fundamental architectural shift. We've also taken
criticism in some quarters, because it was believed that it was
referring to the code being nasty. No, the code isn't nasty, just
the operation itself is rather odd. Fixed for eternity (probably
not). ........ ................
2009-08-28 20:05 +0000 [r214700] Kevin P. Fleming <kpfleming@digium.com>
* makeopts.in, Makefile, /, configure,
include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
214696 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r214696 |
kpfleming | 2009-08-28 15:01:21 -0500 (Fri, 28 Aug 2009) | 9
lines Ensure that CFLAGS and/or LDFLAGS provided to configure
script are preserved. Cross-compilation environments want to
provide 'defaults' for compiler and linker options, and
frequently do this by specifying CFLAGS and LDFLAGS in the
environment or as command-line arguments to the configure script.
This patch modifies the configure script and Makefile to preserve
these settings and ensure they are used in the build process.
........
2009-08-28 18:43 +0000 [r214653] Mark Michelson <mmichelson@digium.com>
* /, include/asterisk/sched.h: Merged revisions 214650 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r214650 | mmichelson | 2009-08-28 13:41:23 -0500 (Fri, 28 Aug
2009) | 3 lines Fix some incorrect documentation of sched_thread
functions. ........
2009-08-27 21:49 +0000 [r214202-214521] Tilghman Lesher <tlesher@digium.com>
* autoconf/libcurl.m4 (added), /, configure,
include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
214518 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r214518 | tilghman | 2009-08-27 16:46:46 -0500 (Thu, 27 Aug 2009)
| 14 lines Merged revisions 214517 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r214517 | tilghman | 2009-08-27 16:45:34 -0500 (Thu, 27 Aug 2009)
| 7 lines Use autoconf to detect libcurl, as this enables
cross-compilation checks, something we didn't allow before.
(closes issue #15714) Reported by: pprindeville Patches:
20090813__issue15714.diff.txt uploaded by tilghman (license 14)
Tested by: pprindeville ........ ................
* main/manager.c, /: Merged revisions 214514 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r214514 |
tilghman | 2009-08-27 16:26:37 -0500 (Thu, 27 Aug 2009) | 7 lines
Ensure that we check for the special value
CONFIG_STATUS_FILEINVALID. (closes issue #15786) Reported by:
a_villacis Patches:
asterisk-1.6.2.0-beta4-manager-fix-crash-on-include-nonexistent-file.patch
uploaded by a villacis (license 660) (Plus a few of my own, to
catch the remaining places within manager.c where it could have
been a problem) ........
* autoconf/ast_ext_lib.m4, /, configure,
include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
214466 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r214466 | tilghman | 2009-08-27 12:28:01 -0500 (Thu, 27 Aug 2009)
| 9 lines Merged revisions 214436 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r214436 | tilghman | 2009-08-27 11:53:58 -0500 (Thu, 27 Aug 2009)
| 2 lines One more build system change, to make the descriptions
look better, if we have better information. ........
................
* autoconf/ast_ext_lib.m4, /, configure,
include/asterisk/autoconfig.h.in: Merged revisions 214360 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r214360 | tilghman | 2009-08-27 11:12:03 -0500
(Thu, 27 Aug 2009) | 10 lines Merged revisions 214357 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r214357 | tilghman | 2009-08-27 11:03:50 -0500 (Thu, 27 Aug 2009)
| 3 lines Make autoheader descriptions render correctly in our
autoconfig.h file. (Figured out while working with issue #14906)
........ ................
* /, channels/chan_sip.c: Merged revisions 214199 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r214199 |
tilghman | 2009-08-26 11:53:03 -0500 (Wed, 26 Aug 2009) | 6 lines
Typo fix ("SIP/2.0 XXX" is 11 chars, not 10) (closes issue
#15362) Reported by: klaus3000 Patches:
chan_sip.c_logmessagefix_patch.txt uploaded by klaus3000 (license
65) ........
2009-08-26 16:39 +0000 [r214196] David Vossel <dvossel@digium.com>
* main/channel.c, /: Merged revisions 214195 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r214195 | dvossel | 2009-08-26 11:38:53 -0500 (Wed, 26 Aug 2009)
| 25 lines Merged revisions 214194 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r214194 | dvossel | 2009-08-26 11:36:42 -0500 (Wed, 26 Aug 2009)
| 19 lines ast_write() ignores ast_audiohook_write() results In
ast_write(), if a channel has a list of audiohooks, those lists
are written to and the resulting frame is what ast_write() should
continue with. The problem was the returned audiohook frame was
not being handled at all, and the original frame passed into it
did not contain the mixed audio, so essentially audio was being
lost. One result of this was chan_spy's whisper mode no longer
worked. To complicate the issue, frames passed into ast_write may
either be a single frame, or a list of frames. So, as the list of
frames is processed in the audiohook_write, the returned frames
had to be added to a new list. (closes issue #15660) Reported by:
corruptor Tested by: dvossel ........ ................
2009-08-25 22:43 +0000 [r213903-214155] Tilghman Lesher <tlesher@digium.com>
* /, configure, include/asterisk/autoconfig.h.in, configure.ac:
Merged revisions 214152 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r214152 |
tilghman | 2009-08-25 17:39:51 -0500 (Tue, 25 Aug 2009) | 4 lines
Not all versions of gnu-linux use glibc, which contains iconv.
Some (especially embedded systems) don't have iconv at all.
(closes issue #15169) Reported by: pprindeville ........
* /, main/say.c: Merged revisions 214071 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r214071 | tilghman | 2009-08-25 14:32:48 -0500 (Tue, 25 Aug 2009)
| 17 lines Merged revisions 214068-214069 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r214068 | tilghman | 2009-08-25 14:26:50 -0500 (Tue, 25 Aug 2009)
| 6 lines Fix pronunciation of German dates. (closes issue
#15273) Reported by: Benjamin Kluck Patches: say_c.patch uploaded
by Benjamin Kluck (license 803) ........ r214069 | tilghman |
2009-08-25 14:28:42 -0500 (Tue, 25 Aug 2009) | 2 lines I should
always compile before committing... ........ ................
* /, pbx/pbx_dundi.c: Merged revisions 213975 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r213975 |
tilghman | 2009-08-25 01:51:12 -0500 (Tue, 25 Aug 2009) | 6 lines
DUNDILOOKUP function in 1.6 should use comma delimiters. (closes
issue #15322) Reported by: chappell Patches:
dundilookup-0015322.patch uploaded by chappell (license 8)
........
* main/pbx.c, /: Merged revisions 213971 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r213971 | tilghman | 2009-08-25 01:35:37 -0500 (Tue, 25 Aug 2009)
| 14 lines Merged revisions 213970 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r213970 | tilghman | 2009-08-25 01:34:44 -0500 (Tue, 25 Aug 2009)
| 7 lines Improve error message by informing user exactly which
function is missing a parethesis. (closes issue #15242) Reported
by: Nick_Lewis Patches: pbx.c-funcparenthesis.patch2 uploaded by
dbrooks (license 790) pbx.c-funcparenthesis-1.4.diff uploaded by
loloski (license 68) ........ ................
* Makefile, /: Merged revisions 213904 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r213904 |
tilghman | 2009-08-24 21:54:07 -0500 (Mon, 24 Aug 2009) | 6 lines
The DTD should be installed in the same path as the rest of the
XML documentation. (closes issue #15344) Reported by: tzafrir
Patches: makefile_appdocs_dtd.diff uploaded by tzafrir (license
46) ........
* Makefile, /: Merged revisions 213900 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r213900 | tilghman | 2009-08-24 21:41:17 -0500 (Mon, 24 Aug 2009)
| 11 lines Merged revisions 213899 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r213899 | tilghman | 2009-08-24 21:40:22 -0500 (Mon, 24 Aug 2009)
| 4 lines Use the default runlevels for Debian derivatives,
instead of making up our own. (closes issue #14730) Reported by:
pkempgen ........ ................
2009-08-24 16:49 +0000 [r213836] Jeff Peeler <jpeeler@digium.com>
* apps/app_voicemail.c, /: Merged revisions 213833 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r213833 | jpeeler | 2009-08-24 11:43:57 -0500 (Mon, 24 Aug 2009)
| 14 lines Fix storage of greetings when using IMAP_STORAGE The
store macro was not getting called preventing storage of IMAP
greetings at all. This has been corrected along with fixing
checking if the imapgreetings option is turned on to store the
greeting in IMAP. Lastly, the attachment filename was incorrectly
using the full path instead of just the basename, which was
causing problems with retrieval of the greeting. (closes issue
#14950) Reported by: noahisaac (closes issue #15729) Reported by:
lmadsen ........
2009-08-24 04:54 +0000 [r213791] Moises Silva <moises.silva@gmail.com>
* channels/chan_dahdi.c, /: Merged revisions 213790 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r213790 | moy | 2009-08-24 00:46:28 -0400 (Mon, 24 Aug 2009) | 1
line improve handling of openr2_chan_disconnect_call API failure,
unlikely, but happened on openr2 library bug ........
2009-08-21 22:54 +0000 [r213739] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_sip.c: Merged revisions 213738 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r213738 |
tilghman | 2009-08-21 17:36:39 -0500 (Fri, 21 Aug 2009) | 2 lines
Clarifying comments in sip_register, and removing a dead section
........
2009-08-21 22:23 +0000 [r213721] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 213716 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r213716 |
dvossel | 2009-08-21 17:22:11 -0500 (Fri, 21 Aug 2009) | 10 lines
Register request line contains wrong address when user domain and
register host differ (closes issue #15539) Reported by:
Nick_Lewis Patches: chan_sip.c-registraraddr.patch uploaded by
Nick (license 657) register_domain_fix_1.6.2 uploaded by dvossel
(license 671) Tested by: Nick_Lewis, dvossel ........
2009-08-21 21:44 +0000 [r213698] Kevin P. Fleming <kpfleming@digium.com>
* apps/app_voicemail.c, /: Merged revisions 213697 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r213697 | kpfleming | 2009-08-21 16:39:51 -0500 (Fri, 21 Aug
2009) | 12 lines Ensure that realtime mailboxes properly report
status on subscription. This patch modifies app_voicemail's
response to mailbox status subscriptions (via the internal event
system) to ensure that a subscription triggers an explicit poll
of the mailbox, so the subscriber can get an immediate cached
event with that status. Previously, the cache was only populated
with the status of non-realtime mailboxes. (closes issue #15717)
Reported by: natmlt ........
2009-08-21 21:12 +0000 [r213636] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 213635 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r213635 |
dvossel | 2009-08-21 16:02:50 -0500 (Fri, 21 Aug 2009) | 5 lines
fixes sip register parsing when user@domain is used (issue
#15008) (issue #15672) ........
2009-08-21 16:55 +0000 [r213563] Tilghman Lesher <tlesher@digium.com>
* include/asterisk.h, /: Merged revisions 213560 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r213560 | tilghman | 2009-08-21 11:53:52 -0500 (Fri, 21 Aug 2009)
| 14 lines Merged revisions 213559 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r213559 | tilghman | 2009-08-21 11:52:53 -0500 (Fri, 21 Aug 2009)
| 7 lines Permit DEBUG_FD_LEAKS to be used with C++ source files.
(closes issue #15698) Reported by: slavon Patches:
20090817__issue15698.diff.txt uploaded by tilghman (license 14)
Tested by: slavon, tilghman ........ ................
2009-08-21 16:06 +0000 [r213497] Jason Parker <jparker@digium.com>
* /, configs/queues.conf.sample: Merged revisions 213494 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r213494 | qwell | 2009-08-21 11:04:21 -0500
(Fri, 21 Aug 2009) | 12 lines Merged revisions 213493 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r213493 | qwell | 2009-08-21 11:03:21 -0500 (Fri, 21 Aug 2009) |
5 lines Clarify queues.conf comments to specify that variables
should be set in the dialplan. (closes issue #15755) Reported by:
trendboy ........ ................
2009-08-21 04:25 +0000 [r213475-213481] Moises Silva <moises.silva@gmail.com>
* channels/chan_dahdi.c, /: Merged revisions 213454 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r213454 | moy | 2009-08-21 00:09:26 -0400 (Fri, 21 Aug 2009) | 1
line increment the mfcr2 monitor count when clearing the call
request ........
* channels/chan_dahdi.c, /: Merged revisions 213216 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r213216 | moy | 2009-08-19 23:26:59 -0400 (Wed, 19 Aug 2009) | 1
line fixed bug caused by calling ast_request without calling
ast_call on an R2 channel, ie, CHANISAVAIL ........
2009-08-21 03:53 +0000 [r213453] Terry Wilson <twilson@digium.com>
* main/loader.c, /: Merged revisions 213450 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r213450 |
twilson | 2009-08-20 22:48:54 -0500 (Thu, 20 Aug 2009) | 2 lines
Make LOAD_ORDER actually work ........
2009-08-20 21:50 +0000 [r213413] Jeff Peeler <jpeeler@digium.com>
* apps/app_voicemail.c, /: Merged revisions 213404 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r213404 | jpeeler | 2009-08-20 16:33:11 -0500 (Thu, 20 Aug 2009)
| 12 lines Fix greeting retrieval from IMAP Properly check for
the current voicemail state and if it doesn't exist, create it.
(closes issue #14597) Reported by: wtca Patches: 14597_v2.patch
uploaded by mmichelson (license 60) Tested by: jpeeler ........
2009-08-20 20:37 +0000 [r213350] Matthew Nicholson <mnicholson@digium.com>
* /, main/features.c: Merged revisions 213327 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r213327 |
mnicholson | 2009-08-20 15:29:32 -0500 (Thu, 20 Aug 2009) | 7
lines Fix a crash by checking the proper pointer for validity
before deferencing it. (closes issue #15751) Reported by: atis
Patches: ast_bridge_call_peer_cdr.patch uploaded by atis (license
242) ........
2009-08-19 22:41 +0000 [r213182] Jason Parker <jparker@digium.com>
* main/alaw.c, main/ulaw.c, /: Merged revisions 213179 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r213179 | qwell | 2009-08-19 17:38:46 -0500 (Wed, 19 Aug 2009) |
5 lines Fix compile when certain G711 menuselect options are
enabled. (closes issue #15697) Reported by: slavon ........
2009-08-19 21:25 +0000 [r213128] David Vossel <dvossel@digium.com>
* apps/app_mixmonitor.c, /: Merged revisions 213113 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r213113 | dvossel | 2009-08-19 16:21:00 -0500
(Wed, 19 Aug 2009) | 14 lines Merged revisions 213103 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r213103 | dvossel | 2009-08-19 16:18:37 -0500 (Wed, 19 Aug 2009)
| 8 lines Fixes memory leak caused by incorrectly freeing
mixmonitor (closes issue #15699) Reported by: edantie Patches:
mixmonitor.patch uploaded by edantie (license 862) ........
................
2009-08-19 21:22 +0000 [r213095-213117] Tilghman Lesher <tlesher@digium.com>
* configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions
213098 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r213098 |
tilghman | 2009-08-19 16:05:17 -0500 (Wed, 19 Aug 2009) | 9 lines
Better parsing for the "register" line Allows characters that are
otherwise used as delimiters to be used within certain fields
(like the secret). (closes issue #15008, closes issue #15672)
Reported by: tilghman Patches: 20090818__issue15008.diff.txt
uploaded by tilghman (license 14) Tested by: lmadsen, tilghman
........
* /, channels/chan_sip.c: Merged revisions 213093 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r213093 |
tilghman | 2009-08-19 15:29:41 -0500 (Wed, 19 Aug 2009) | 7 lines
If we have realtime caching enabled, 'sip reload' must purge
users/peers, even if the config files haven't changed. (closes
issue #12869) Reported by: bcnit Patches:
20090819__issue12869__2.diff.txt uploaded by tilghman (license
14) Tested by: lasko ........
2009-08-19 15:35 +0000 [r213047] Russell Bryant <russell@digium.com>
* /, main/features.c: Merged revisions 213046 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r213046 |
russell | 2009-08-19 10:32:18 -0500 (Wed, 19 Aug 2009) | 4 lines
Don't blow up on a NULL cdr. Reported in #asterisk-dev. ........
2009-08-18 20:34 +0000 [r212931-212944] Kevin P. Fleming <kpfleming@digium.com>
* /: Merged revisions 212939 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r212939 |
kpfleming | 2009-08-18 15:33:34 -0500 (Tue, 18 Aug 2009) | 1 line
Remove some accidentally-committed properties. ........
* sounds/Makefile, doc/tex/asterisk.tex, CREDITS, /,
UPGRADE-1.4.txt, sounds/sounds.xml, build_tools/prep_tarball:
Merged revisions 212922 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r212922 |
kpfleming | 2009-08-18 15:29:37 -0500 (Tue, 18 Aug 2009) | 6
lines Convert this branch to Opsound music-on-hold. For more
details:
http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/
........
2009-08-18 19:28 +0000 [r212866] Tilghman Lesher <tlesher@digium.com>
* /, configs/extconfig.conf.sample: Merged revisions 212857 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r212857 | tilghman | 2009-08-18 14:25:09 -0500 (Tue, 18
Aug 2009) | 4 lines Make the default extconfig.conf match entries
with the sample res_mysql.conf. This eliminates a future source
of possible confusion with the configuration of 1.6.1 and higher.
........
2009-08-18 16:56 +0000 [r212769] Richard Mudgett <rmudgett@digium.com>
* channels/misdn/isdn_lib.c, /: Merged revisions 212758 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r212758 | rmudgett | 2009-08-18 11:29:47 -0500
(Tue, 18 Aug 2009) | 9 lines Merged revisions 212727 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r212727 | rmudgett | 2009-08-18 11:00:56 -0500 (Tue, 18
Aug 2009) | 1 line Removed some deadwood and added some doxygen
comments. ........ ................
2009-08-18 16:41 +0000 [r212767] Sean Bright <sean@malleable.com>
* main/manager.c, /: Merged revisions 212764 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r212764 | seanbright | 2009-08-18 12:38:36 -0400 (Tue, 18 Aug
2009) | 18 lines Merged revisions 212763 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r212763 | seanbright | 2009-08-18 12:36:00 -0400 (Tue, 18 Aug
2009) | 11 lines Delay the creation of temporary files until we
have a valid manager command to handle. Without this patch,
asterisk creates a temporary file before determining if the
specified command is valid. If invalid, we weren't properly
cleaning up the file. (closes issue #15730) Reported by: zmehmood
Patches: M15730.diff uploaded by junky (license 177) Tested by:
zmehmood ........ ................
2009-08-17 20:01 +0000 [r212631] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c, /: Merged revisions 212627 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r212627 | tilghman | 2009-08-17 14:57:42 -0500 (Mon, 17 Aug 2009)
| 4 lines Check the return value of opendir(3), or we may crash.
(closes issue #15720) Reported by: tobias_e ........
2009-08-17 18:56 +0000 [r212580-212584] Sean Bright <sean@malleable.com>
* /, channels/chan_agent.c: Merged revisions 212581 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r212581 | seanbright | 2009-08-17 14:50:24 -0400 (Mon, 17 Aug
2009) | 5 lines Correct spelling of AGENTACCEPTDTMF in
chan_agent. (closes issue #15668) Reported by: davidw ........
* main/logger.c: Merged revisions 212574 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r212574 |
seanbright | 2009-08-17 14:18:16 -0400 (Mon, 17 Aug 2009) | 8
lines Correct the return value check for ast_safe_system. The
logic here was reversed as ast_safe_system returns -1 on error
and not on success. Fix suggested by reporter. (closes issue
#15667) Reported by: loic ........
2009-08-17 16:52 +0000 [r212509] Jeff Peeler <jpeeler@digium.com>
* channels/misdn_config.c, /: Merged revisions 212506 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r212506 | jpeeler | 2009-08-17 11:50:45 -0500
(Mon, 17 Aug 2009) | 19 lines Merged revisions 212498 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r212498 | jpeeler | 2009-08-17 11:34:56 -0500 (Mon, 17 Aug 2009)
| 12 lines Fix segfault when reloading chan_misdn. If more ports
were specified than configured in misdn.conf a reload would crash
asterisk. The problem was the unconfigured port was using data
from the previously configured port. When the data for an
unconfigured port was freed a crash would result from the double
free. (closes issue #12113) Reported by: agupta Patches:
bug12113.patch uploaded by jpeeler (license 325) ........
................
2009-08-17 15:51 +0000 [r212434] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 212431 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r212431 | rmudgett | 2009-08-17 10:42:51 -0500
(Mon, 17 Aug 2009) | 16 lines Merged revisions 212430 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 Fix
uninitialized variable causing random MWI indications. (closes
issue #15727) Reported by: doda Patches: dahdi_changes.patch
uploaded by doda (license 853) ........ r212430 | rmudgett |
2009-08-17 10:36:28 -0500 (Mon, 17 Aug 2009) | 1 line Fix
uninitialized variable. ........ ................
2009-08-14 17:37 +0000 [r212250] Tilghman Lesher <tlesher@digium.com>
* funcs/func_curl.c, /: Merged revisions 212249 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r212249 |
tilghman | 2009-08-14 12:36:40 -0500 (Fri, 14 Aug 2009) | 2 lines
Add SSL_VERIFYPEER, as requested on the -users list ........
2009-08-13 15:47 +0000 [r212116] Kevin P. Fleming <kpfleming@digium.com>
* /, channels/chan_sip.c: Merged revisions 212113 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r212113 |
kpfleming | 2009-08-13 10:46:25 -0500 (Thu, 13 Aug 2009) | 3
lines Ensure that T38FaxVersion is put into outgoing SDP in the
proper case. ........
2009-08-13 13:56 +0000 [r212070] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 212067 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r212067 |
file | 2009-08-13 10:51:04 -0300 (Thu, 13 Aug 2009) | 2 lines
Check an actual populated variable when seeing if we need to do
video or not. ........
2009-08-13 11:47 +0000 [r212030] Gavin Henry <ghenry@suretecsystems.com>
* contrib/scripts/asterisk.ldap-schema,
contrib/scripts/asterisk.ldif, /: Merged revisions 212027 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r212027 | ghenry | 2009-08-13 12:37:12 +0100 (Thu, 13
Aug 2009) | 6 lines Fixed typo (closes issue #15710) Reported by:
suretec ........
2009-08-12 23:16 +0000 [r211951-211959] Matthew Nicholson <mnicholson@digium.com>
* apps/app_queue.c, /: Merged revisions 211957 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r211957 | mnicholson | 2009-08-12 18:14:36 -0500 (Wed, 12 Aug
2009) | 17 lines Merged revisions 211953 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r211953 | mnicholson | 2009-08-12 18:04:02 -0500 (Wed, 12 Aug
2009) | 10 lines This patch adds additional checking when
generating queue log TRANSFER events. The additional checks
prevent generation of false TRANSFER events in certain
situations. (closes issue #14536) Reported by: aragon Patches:
queue-log-xfer-fix1.diff uploaded by mnicholson (license 96)
Tested by: aragon, mnicholson ........ ................
* /, channels/chan_sip.c: Merged revisions 211876 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r211876 |
mnicholson | 2009-08-12 14:53:14 -0500 (Wed, 12 Aug 2009) | 11
lines Make asterisk handle 423 Interval Too Short messages
better. This change uses separate values for the acceptable
minimum expiry provided by the 423 error and the expiry value
stored in the configuration file. Previously, the value pulled
from the configuration file would be overwritten. (closes issue
#14366) Reported by: Nick_Lewis Patches: sip-expiry-fix1.diff
uploaded by mnicholson (license 96) chan_sip.c-reqexpiry.patch
uploaded by Nick (license 657) Tested by: mnicholson ........
2009-08-12 16:21 +0000 [r211785] Gavin Henry <ghenry@suretecsystems.com>
* res/res_config_ldap.c, contrib/scripts/asterisk.ldap-schema,
contrib/scripts/asterisk.ldif, /: Merged revisions 211767 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r211767 | ghenry | 2009-08-12 17:00:46 +0100 (Wed, 12
Aug 2009) | 33 lines Added three new attributes and applied a
patch to res_config_ldap.c attributetype (
AstAccountSubscribeContext NAME 'AstAccountSubscribeContext' DESC
'Asterisk subscribe context' EQUALITY caseIgnoreMatch SUBSTR
caseIgnoreSubstringsMatch SYNTAX 1.3.6.1.4.1.1466.115.121.1.15)
attributetype ( AstAccountIpAddr NAME 'AstAccountIpAddr' DESC
'Asterisk aaccount IP address' EQUALITY caseIgnoreMatch SUBSTR
caseIgnoreSubstringsMatch SYNTAX 1.3.6.1.4.1.1466.115.121.1.15)
attributetype ( AstAccountUserAgent NAME 'AstAccountUserAgent'
DESC 'Asterisk account user context' EQUALITY caseIgnoreMatch
SUBSTR caseIgnoreSubstringsMatch SYNTAX
1.3.6.1.4.1.1466.115.121.1.15) and patch
fix_empty_attributes_1.6.1.4_v2.patch (closes issue #13725)
Reported by: macogeek Patches:
fix_empty_attributes_1.6.1.4_v2.patch uploaded by xvisor (license
863) Tested by: suretec ........
2009-08-10 19:51 +0000 [r211580-211585] Tilghman Lesher <tlesher@digium.com>
* doc/CODING-GUIDELINES, /: Merged revisions 211584 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r211584 | tilghman | 2009-08-10 14:49:41 -0500
(Mon, 10 Aug 2009) | 9 lines Merged revisions 211583 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r211583 | tilghman | 2009-08-10 14:48:48 -0500 (Mon, 10
Aug 2009) | 1 line Conversion specifiers, not format specifiers
........ ................
* apps/app_queue.c, apps/app_talkdetect.c, agi/eagi-sphinx-test.c,
res/res_config_curl.c, channels/chan_usbradio.c,
channels/chan_misdn.c, res/snmp/agent.c, apps/app_sms.c,
apps/app_verbose.c, apps/app_stack.c, apps/app_mixmonitor.c,
main/asterisk.c, main/dsp.c, main/timing.c,
doc/CODING-GUIDELINES, funcs/func_speex.c, main/frame.c,
utils/muted.c, apps/app_meetme.c, apps/app_alarmreceiver.c,
cdr/cdr_pgsql.c, res/res_musiconhold.c, channels/chan_iax2.c,
apps/app_followme.c, main/enum.c, main/indications.c,
res/res_config_sqlite.c, channels/misdn_config.c, utils/frame.c,
main/cli.c, pbx/pbx_loopback.c, channels/chan_phone.c,
funcs/func_enum.c, res/res_smdi.c, channels/chan_skinny.c,
funcs/func_odbc.c, apps/app_minivm.c, res/res_agi.c,
res/res_config_ldap.c, apps/app_adsiprog.c,
funcs/func_dialplan.c, main/pbx.c, main/dnsmgr.c,
funcs/func_sprintf.c, funcs/func_timeout.c, channels/chan_sip.c,
apps/app_privacy.c, res/res_limit.c, apps/app_waitforsilence.c,
codecs/codec_speex.c, agi/eagi-test.c, apps/app_morsecode.c,
funcs/func_cut.c, channels/chan_oss.c, main/netsock.c,
apps/app_waitforring.c, funcs/func_channel.c, apps/app_macro.c,
pbx/pbx_dundi.c, utils/extconf.c, pbx/pbx_config.c,
apps/app_chanspy.c, res/res_odbc.c, apps/app_voicemail.c,
apps/app_dahdibarge.c, funcs/func_rand.c, apps/app_readfile.c, /,
apps/app_record.c, main/utils.c, cdr/cdr_adaptive_odbc.c,
res/res_http_post.c, main/config.c, res/ael/pval.c, main/cdr.c,
main/channel.c, channels/chan_dahdi.c, pbx/pbx_spool.c,
main/manager.c, apps/app_setcallerid.c, apps/app_osplookup.c,
main/features.c, main/http.c, channels/xpmr/xpmr.c,
apps/app_rpt.c, channels/chan_mgcp.c, res/res_config_pgsql.c,
channels/chan_agent.c, funcs/func_math.c, apps/app_waituntil.c,
apps/app_disa.c, main/acl.c, apps/app_originate.c,
channels/iax2-provision.c: AST-2009-005
2009-08-10 14:15 +0000 [r211350] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 211347 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r211347 |
file | 2009-08-10 11:07:44 -0300 (Mon, 10 Aug 2009) | 5 lines Fix
retrieval of the port used for the video stream when adding SDP
to a SIP message. (closes issue #15121) Reported by: jsmith
........
2009-08-09 15:43 +0000 [r211235-211278] Tilghman Lesher <tlesher@digium.com>
* /, main/astfd.c: Merged revisions 211275 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r211275 | tilghman | 2009-08-09 10:42:02 -0500 (Sun, 09 Aug 2009)
| 9 lines Merged revisions 211274 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r211274 | tilghman | 2009-08-09 10:41:01 -0500 (Sun, 09 Aug 2009)
| 2 lines Small oops. Clear the flags which have been checked.
........ ................
* apps/app_stack.c, /: Merged revisions 211232 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r211232 |
tilghman | 2009-08-09 02:11:22 -0500 (Sun, 09 Aug 2009) | 4 lines
Check for NULL frame, before dereferencing pointer. (closes issue
#15617) Reported by: rain ........
2009-08-07 20:18 +0000 [r211122] Russell Bryant <russell@digium.com>
* apps/app_chanspy.c, /: Merged revisions 211113 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r211113 | russell | 2009-08-07 15:12:21 -0500 (Fri, 07 Aug 2009)
| 11 lines Recorded merge of revisions 211112 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r211112 | russell | 2009-08-07 15:11:31 -0500 (Fri, 07 Aug 2009)
| 4 lines Resolve a deadlock involving app_chanspy and
masquerades. (ABE-1936) ........ ................
2009-08-07 18:20 +0000 [r211051] Tilghman Lesher <tlesher@digium.com>
* apps/app_queue.c, /: Merged revisions 211040 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r211040 | tilghman | 2009-08-07 13:17:41 -0500 (Fri, 07 Aug 2009)
| 21 lines Merged revisions 211038 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r211038 | tilghman | 2009-08-07 13:16:28 -0500 (Fri, 07 Aug 2009)
| 14 lines QUEUE_MEMBER_LIST _really_ wants the interface name,
not the membername. This is a partial revert of revision 82590,
which was an attempted cleanup, but in reality, it broke
QUEUE_MEMBER_LIST, which has always been intended as a method by
which component interfaces could be queried from the queue.
Membername isn't useful here, because that field cannot be used
to obtain further information about the member. See the
documentation on QUEUE_MEMBER_LIST, RemoveQueueMember,
QUEUE_MEMBER_PENALTY, and the various AMI commands which take a
member argument for further justification. (closes issue #15664)
Reported by: rain Patches: app_queue-queue_member_list.diff
uploaded by rain (license 327) ........ ................
2009-08-07 13:10 +0000 [r210995] Kevin P. Fleming <kpfleming@digium.com>
* main/udptl.c, /: Merged revisions 210992 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r210992 |
kpfleming | 2009-08-07 08:08:00 -0500 (Fri, 07 Aug 2009) | 13
lines Workaround broken T.38 endpoints that offer tiny
MaxDatagram sizes. Some T.38 endpoints treat T38FaxMaxDatagram as
the maximum IFP size that should be sent to them, rather than the
maximum packet payload size. If such an endpoint also requests
UDPRedundancy as the error correction mode, we'll end up
calculating a tiny maximum IFP size, so small as to be unusable.
This patch sets a lower bound on what we'll consider the remote's
maximum IFP size to be, assuming that endpoints that do this
really can accept larger packets than they've offered to accept.
(closes issue #15649) Reported by: dazza76 ........
2009-08-06 21:47 +0000 [r210911-210917] Tilghman Lesher <tlesher@digium.com>
* main/channel.c, /: Merged revisions 210914 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r210914 | tilghman | 2009-08-06 16:46:01 -0500 (Thu, 06 Aug 2009)
| 14 lines Merged revisions 210913 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r210913 | tilghman | 2009-08-06 16:45:01 -0500 (Thu, 06 Aug 2009)
| 7 lines Because channel information can be accessed outside of
the channel thread, we must lock the channel prior to modifying
it. (closes issue #15397) Reported by: caspy Patches:
20090714__issue15397.diff.txt uploaded by tilghman (license 14)
Tested by: caspy ........ ................
* apps/app_stack.c, include/asterisk/app.h, /, main/app.c: Merged
revisions 210908 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r210908 |
tilghman | 2009-08-06 16:29:26 -0500 (Thu, 06 Aug 2009) | 9 lines
Allow Gosub to recognize quote delimiters without consuming them.
(closes issue #15557) Reported by: rain Patches:
20090723__issue15557.diff.txt uploaded by tilghman (license 14)
Tested by: rain Review: https://reviewboard.asterisk.org/r/316/
........
2009-08-06 17:49 +0000 [r210820] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 210817 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r210817 |
file | 2009-08-06 14:47:04 -0300 (Thu, 06 Aug 2009) | 11 lines
Accept additional T.38 reinvites after an initial one has been
handled. Discussion of this subject has yielded that it is not
actually acceptable to change T.38 parameters after the initial
reinvite but declining is harsh and can cause the fax to fail
when it may be possible to allow it to continue. This patch
changes things so that additional T.38 reinvites are accepted but
parameter changes ignored. This gives the fax a fighting chance.
(closes issue #15610) Reported by: huangtx2009 ........
2009-08-05 20:43 +0000 [r210686] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 210640 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r210640 | rmudgett | 2009-08-05 14:40:03 -0500
(Wed, 05 Aug 2009) | 21 lines Merged revisions 210575 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r210575 | rmudgett | 2009-08-05 14:18:56 -0500 (Wed, 05 Aug 2009)
| 14 lines Dialplan starts execution before the channel setup is
complete. * Issue 15655: For the case where dialing is complete
for an incoming call, dahdi_new() was asked to start the PBX and
then the code set more channel variables. If the dialplan hungup
before these channel variables got set, asterisk would likely
crash. * Fixed potential for overlap incoming call to erroneously
set channel variables as global dialplan variables if the
ast_channel structure failed to get allocated. * Added missing
set of CALLINGSUBADDR in the dialing is complete case. (closes
issue #15655) Reported by: alecdavis ........ ................
2009-08-05 18:56 +0000 [r210565-210566] Leif Madsen <lmadsen@digium.com>
* /: Merged revisions 210564 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r210564 | lmadsen | 2009-08-05 13:49:58 -0500 (Wed, 05 Aug 2009)
| 19 lines Merged revisions 210563 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r210563 | lmadsen | 2009-08-05 13:46:21 -0500 (Wed, 05 Aug 2009)
| 11 lines Update imapstorage.txt documentation. Updated the
imapstorage.txt documentation to reflect that issues with
c-client versions older than 2007 seem to cause crashing issues
that are not seen with more recent versions. Documentation has
been updated to reflect this. (closes issue #14496) Reported by:
vbcrlfuser Patches: __20090727-imap-documentation-patch.txt
uploaded by lmadsen (license 10) Tested by: lmadsen, mmichelson,
dbrooks ........ ................
* doc/tex/imapstorage.tex: Merged revisions 210564 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r210564 | lmadsen | 2009-08-05 13:49:58 -0500
(Wed, 05 Aug 2009) | 19 lines Merged revisions 210563 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r210563 | lmadsen | 2009-08-05 13:46:21 -0500 (Wed, 05 Aug 2009)
| 11 lines Update imapstorage.txt documentation. Updated the
imapstorage.txt documentation to reflect that issues with
c-client versions older than 2007 seem to cause crashing issues
that are not seen with more recent versions. Documentation has
been updated to reflect this. (closes issue #14496) Reported by:
vbcrlfuser Patches: __20090727-imap-documentation-patch.txt
uploaded by lmadsen (license 10) Tested by: lmadsen, mmichelson,
dbrooks ........ ................
2009-08-04 14:55 +0000 [r210191-210241] Kevin P. Fleming <kpfleming@digium.com>
* Makefile, /: Merged revisions 210238 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r210238 | kpfleming | 2009-08-04 09:53:00 -0500 (Tue, 04 Aug
2009) | 16 lines Merged revisions 210237 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r210237 | kpfleming | 2009-08-04 09:51:39 -0500 (Tue, 04 Aug
2009) | 10 lines Eliminate spurious compiler warnings from system
headers on *BSD platforms. Ensure that system headers located in
/usr/local/include are actually treated as system headers by the
compiler, and not as local headers which are subject to warnings
from the -Wundef compiler option and others. (closes issue
#15606) Reported by: mvanbaak ........ ................
* configs/sip.conf.sample, configs/skinny.conf.sample, main/rtp.c,
channels/chan_mgcp.c, doc/chan_sip-perf-testing.txt,
contrib/scripts/realtime_pgsql.sql, /, channels/chan_sip.c,
channels/chan_skinny.c, configs/mgcp.conf.sample,
doc/res_config_sqlite.txt, doc/tex/phoneprov.tex, UPGRADE.txt,
configs/res_ldap.conf.sample: Merged revisions 210190 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r210190 | kpfleming | 2009-08-03 15:48:48 -0500 (Mon, 03
Aug 2009) | 11 lines Rename 'canreinvite' option to
'directmedia', with backwards compatibility. It is clear from
multiple mailing list, forum, wiki and other sorts of posts that
users don't really understand the effects that the 'canreinvite'
config option actually has, and that in some cases they think
that setting it to 'no' will actually cause various other
features (T.38, MOH, etc.) to not work properly, when in fact
this is not the case. This patch changes the proper name of the
option to what it should have been from the beginning
('directmedia'), but preserves backwards compatibility for
existing configurations. ........
2009-08-01 11:33 +0000 [r209837-209906] Russell Bryant <russell@digium.com>
* main/db1-ast/mpool/mpool.c, /: Merged revisions 209887 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r209887 | russell | 2009-08-01 06:29:25 -0500
(Sat, 01 Aug 2009) | 12 lines Merged revisions 209879 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r209879 | russell | 2009-08-01 06:27:25 -0500 (Sat, 01 Aug 2009)
| 5 lines Resolve a valgrind warning about a read from
uninitialized memory. (issue #15396) Reported by: aragon ........
................
* apps/app_milliwatt.c, /: Merged revisions 209839 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r209839 | russell | 2009-08-01 06:02:07 -0500
(Sat, 01 Aug 2009) | 20 lines Merged revisions 209838 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r209838 | russell | 2009-08-01 05:59:05 -0500 (Sat, 01 Aug 2009)
| 13 lines Modify how Playtones() is used in Milliwatt() to
resolve gain issue. When Milliwatt() was changed internally to
use Playtones() so that the proper tone was used, it introduced a
drop in gain in the output signal. So, use the playtones API
directly and specify a volume argument such that the output
matches the gain of the original Milliwatt() code. (closes issue
#15386) Reported by: rue_mohr Patches: issue_15386.rev2.diff
uploaded by russell (license 2) Tested by: rue_mohr ........
................
* /, main/event.c: Merged revisions 209835 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r209835 |
russell | 2009-08-01 05:43:40 -0500 (Sat, 01 Aug 2009) | 6 lines
Fix ast_event_queue_and_cache() to actually do the cache() part.
(closes issue #15624) Reported by: ffossard Tested by: russell
........
2009-08-01 01:34 +0000 [r209816] Kevin P. Fleming <kpfleming@digium.com>
* pbx/pbx_config.c, channels/misdn/isdn_lib.c, utils/frame.c,
main/pbx.c, /, main/Makefile, channels/misdn/ie.c: Merged
revisions 209760-209761 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r209760 | kpfleming | 2009-07-31 20:03:07 -0500 (Fri, 31 Jul
2009) | 13 lines Merged revisions 209759 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r209759 | kpfleming | 2009-07-31 19:52:00 -0500 (Fri, 31 Jul
2009) | 7 lines Minor changes inspired by testing with latest
GCC. The latest GCC (what will become 4.5.x) has a few new
warnings, that in these cases found some either downright buggy
code, or at least seriously poorly designed code that could be
improved. ........ ................ r209761 | kpfleming |
2009-07-31 20:04:06 -0500 (Fri, 31 Jul 2009) | 1 line Revert
accidental Makefile change. ................
2009-07-31 22:01 +0000 [r209715] Russell Bryant <russell@digium.com>
* /, main/event.c: Merged revisions 209711 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r209711 |
russell | 2009-07-31 16:53:31 -0500 (Fri, 31 Jul 2009) | 2 lines
Fix some places where ast_event_type was used instead of
ast_event_ie_type. ........
2009-07-30 18:51 +0000 [r209594] David Brooks <dbrooks@digium.com>
* channels/chan_console.c, include/asterisk/abstract_jb.h,
apps/app_forkcdr.c, channels/chan_dahdi.c,
contrib/init.d/rc.debian.asterisk, /, apps/app_sms.c,
codecs/lpc10/pitsyn.c: Merged revisions 209554 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r209554 |
dbrooks | 2009-07-30 11:07:05 -0500 (Thu, 30 Jul 2009) | 6 lines
Fixes numerous spelling errors. Patch submitted by alecdavis.
(closes issue #15595) Reported by: alecdavis ........
2009-07-30 14:40 +0000 [r209518] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 209516 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r209516 |
mmichelson | 2009-07-30 09:38:21 -0500 (Thu, 30 Jul 2009) | 8
lines Fix a crash that can result if text codecs are allowed but
textsupport is disabled. (closes issue #15596) Reported by:
fabled Patches: sip-red.patch uploaded by fabled (license 448)
........
2009-07-28 Leif Madsen <lmadsen@digium.com>
* Release Asterisk 1.6.2.0-beta4
2009-07-28 00:19 +0000 [r209328] Tilghman Lesher <tlesher@digium.com>
* /, sounds/sounds.xml: Merged revisions 209317 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r209317 | tilghman | 2009-07-27 19:14:12 -0500 (Mon, 27 Jul 2009)
| 9 lines Merged revisions 209315 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r209315 | tilghman | 2009-07-27 19:12:03 -0500 (Mon, 27 Jul 2009)
| 2 lines Publish French extra sounds ........ ................
2009-07-27 21:44 +0000 [r209265-209282] Kevin P. Fleming <kpfleming@digium.com>
* /, apps/app_fax.c: Merged revisions 209279 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r209279 |
kpfleming | 2009-07-27 16:43:36 -0500 (Mon, 27 Jul 2009) | 7
lines Cleanup T.38 negotiation changes. Convert LOG_NOTICE
messages about T.38 negotiation in debug level 1 messages, clean
up some looping logic, and correct an improper use of ast_free()
for freeing an ast_frame. ........
* /, apps/app_fax.c: Merged revisions 209256 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r209256 |
kpfleming | 2009-07-27 16:21:43 -0500 (Mon, 27 Jul 2009) | 10
lines Make T.38 switchover in ReceiveFAX synchronous. In receive
mode, if the channel that ReceiveFAX is running on supports T.38,
we should *always* attempt to switch T.38, rather than listening
for an incoming CNG tone and only triggering on that. The channel
may be using a low-bitrate codec that distorts the CNG tone, the
sending FAX endpoint may not send CNG at all, or there could be a
variety of other reasons that we don't detect it, but in all
those cases if T.38 is available we certainly want to use it.
........
2009-07-27 20:58 +0000 [r209238] Mark Michelson <mmichelson@digium.com>
* main/rtp.c, /: Merged revisions 209235 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r209235 |
mmichelson | 2009-07-27 15:54:54 -0500 (Mon, 27 Jul 2009) | 5
lines Gracefully handle malformed RTP text packets. AST-2009-004
........
2009-07-27 20:33 +0000 [r209234] David Brooks <dbrooks@digium.com>
* res/res_jabber.c, main/loader.c, channels/chan_dahdi.c,
channels/chan_vpb.cc, res/res_smdi.c, /,
include/asterisk/module.h, main/features.c, res/res_agi.c: Merged
revisions 209098 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r209098 |
dbrooks | 2009-07-27 11:33:50 -0500 (Mon, 27 Jul 2009) | 6 lines
Fixing typos. Replaces "recieved" with "received" and "initilize"
with "initialize" (closes issue #15571) Reported by: alecdavis
........
2009-07-27 20:23 +0000 [r209135-209222] Mark Michelson <mmichelson@digium.com>
* res/res_musiconhold.c, /: Merged revisions 209197 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r209197 | mmichelson | 2009-07-27 15:11:42 -0500 (Mon, 27 Jul
2009) | 9 lines Honor channel's music class when using realtime
music on hold. (closes issue #15051) Reported by: alexh Patches:
15051.patch uploaded by mmichelson (license 60) Tested by: alexh
........
* main/udptl.c, /, configs/udptl.conf.sample: Merged revisions
209132 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r209132 | mmichelson | 2009-07-27 12:50:04 -0500 (Mon, 27 Jul
2009) | 24 lines Merged revisions 209131 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r209131 | mmichelson | 2009-07-27 12:44:06 -0500 (Mon, 27 Jul
2009) | 18 lines Allow for UDPTL to use only even-numbered ports
if desired. There are some VoIP providers out there that will not
accept SDP offers with odd numbered UDPTL ports. While it is my
personal opinion that these VoIP providers are misinterpreting
RFC 2327, it really is not a big deal to play along with their
silly little games. Of course, since restricting UDPTL ports to
only even numbers reduces the range of available ports by half,
so the option to use only even port numbers is off by default. A
user can enable the behavior by setting use_even_ports=yes in
udptl.conf. (closes issue #15182) Reported by: CGMChris Patches:
15182.patch uploaded by mmichelson (license 60) Tested by:
CGMChris ........ ................
2009-07-27 16:07 +0000 [r209063] David Brooks <dbrooks@digium.com>
* apps/app_rpt.c, res/res_smdi.c, pbx/pbx_dundi.c: Just replacing
typos "recieved" with "received". From issue #15360, forgot to
apply to trunk and other branches.
2009-07-27 15:40 +0000 [r209059] Kevin P. Fleming <kpfleming@digium.com>
* Makefile, /: Merged revisions 209056 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r209056 |
kpfleming | 2009-07-27 10:38:59 -0500 (Mon, 27 Jul 2009) | 10
lines Restore explicit export of ASTCFLAGS/ASTLDFLAGS and
underscore-variants to sub-makes. During the recent Makefile
improvements I made, it seemed the 'make' was automatically
carrying down the ASTCFLAGS/ASTLDFLAGS settings to sub-makes, so
I removed the explict export of them. However, there are some
circumstances where make does this, and some where it does not,
so I've brought them back to ensure they are always exported. I
also removed an extraneous double setting of _ASTLDFLAGS on *BSD
platforms. ........
2009-07-27 01:23 +0000 [r208927] Jeff Peeler <jpeeler@digium.com>
* channels/chan_iax2.c, /, main/translate.c: Merged revisions
208924 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r208924 | jpeeler | 2009-07-26 20:20:37 -0500 (Sun, 26 Jul 2009)
| 9 lines Merged revisions 208923 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r208923 | jpeeler | 2009-07-26 20:18:31 -0500 (Sun, 26 Jul 2009)
| 2 lines Fix logic errors from 208746 ........ ................
2009-07-26 14:07 +0000 [r208889] Michiel van Baak <michiel@vanbaak.info>
* contrib/scripts/install_prereq, /: Merged revisions 208886 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r208886 | mvanbaak | 2009-07-26 16:00:52 +0200 (Sun, 26
Jul 2009) | 2 lines add OpenBSD to the install_prereq script
........
2009-07-25 12:31 +0000 [r208816-208853] Michiel van Baak <michiel@vanbaak.info>
* contrib/scripts/install_prereq, /: Merged revisions 208848 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r208848 | mvanbaak | 2009-07-25 14:28:38 +0200 (Sat, 25
Jul 2009) | 2 lines libxml2-dev is needed as well by default.
........
* main/cli.c, /, configs/cli_aliases.conf.sample: Merged revisions
208813 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r208813 |
mvanbaak | 2009-07-25 14:03:25 +0200 (Sat, 25 Jul 2009) | 10
lines add default alias reload to run module reload. Requiring
'module reload' to reload everything, including core etc makes
russell very unhappy. The default configuration already loads the
'friendly' aliases template. Added 'reload=module reload' to that
template. Also removed the comment in main/cli.c that reload
should come back. ........
2009-07-25 06:26 +0000 [r208755] Jeff Peeler <jpeeler@digium.com>
* channels/chan_iax2.c, /, channels/chan_skinny.c,
main/translate.c: Merged revisions 208749 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r208749 | jpeeler | 2009-07-25 01:23:18 -0500 (Sat, 25 Jul 2009)
| 13 lines Merged revisions 208746 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r208746 | jpeeler | 2009-07-25 01:19:50 -0500 (Sat, 25 Jul 2009)
| 7 lines Fix compiling under dev-mode with gcc 4.4.0. Mostly
trivial changes, but I did not know of any other way to fix the
"dereferencing type-punned pointer will break strict-aliasing
rules" error without creating a tmp variable in chan_skinny.
........ ................
2009-07-24 21:13 +0000 [r208695-208710] Russell Bryant <russell@digium.com>
* /, pbx/pbx_dundi.c: Merged revisions 208709 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r208709 |
russell | 2009-07-24 16:12:43 -0500 (Fri, 24 Jul 2009) | 2 lines
Remove trailing whitespace. ........
* main/cli.c, /: Merged revisions 208706 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r208706 |
russell | 2009-07-24 15:54:37 -0500 (Fri, 24 Jul 2009) | 6 lines
Note that "reload" needs to be added back. I keep getting annoyed
at having to type "module reload" to reload everything, so I'm
adding a note that we need to add "reload" back. "module reload"
doesn't really make sense as the command to reload everything,
including the core. ........
* main/cli.c, /: Merged revisions 208693 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r208693 |
russell | 2009-07-24 15:25:23 -0500 (Fri, 24 Jul 2009) | 2 lines
Don't log a warning for something that does not affect operation.
........
2009-07-24 19:42 +0000 [r208664] Mark Michelson <mmichelson@digium.com>
* /: Fixing trunk-blocked property.
2009-07-24 18:56 +0000 [r208596] Russell Bryant <russell@digium.com>
* apps/app_dial.c, /: Merged revisions 208593 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r208593 | russell | 2009-07-24 13:42:32 -0500 (Fri, 24 Jul 2009)
| 14 lines Merged revisions 208592 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r208592 | russell | 2009-07-24 13:38:24 -0500 (Fri, 24 Jul 2009)
| 7 lines Do not log an ERROR if autoservice_stop() returns -1.
This does not indicate an error. A return of -1 just means that
the channel has been hung up. (reported in #asterisk-dev)
........ ................
2009-07-24 18:32 +0000 [r208591] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 208588 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r208588 | mmichelson | 2009-07-24 13:31:04 -0500 (Fri, 24 Jul
2009) | 16 lines Merged revisions 208587 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r208587 | mmichelson | 2009-07-24 13:26:50 -0500 (Fri, 24 Jul
2009) | 10 lines Only send a BYE when hanging up a channel that
is up. For cases where Asterisk sends an INVITE and receives a
non 2XX final response, Asterisk would follow the INVITE
transaction by immediately sending a BYE, which was unnecessary.
(closes issue #14575) Reported by: chris-mac ........
................
2009-07-24 15:06 +0000 [r208551] Kevin P. Fleming <kpfleming@digium.com>
* main/udptl.c, /, channels/chan_sip.c, include/asterisk/udptl.h:
Merged revisions 208548 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r208548 |
kpfleming | 2009-07-24 10:02:53 -0500 (Fri, 24 Jul 2009) | 8
lines Resolve a T.38 negotiation issue left over from the
udptl-updates merge. The udptl-updates branch that was merged
yesterday failed to properly send back T.38 SDP responses with
the correct error correction mode, if the incoming SDP from the
other end caused us to change error correction modes. This patch
corrects that situation. ........
2009-07-24 14:39 +0000 [r208545] Michiel van Baak <michiel@vanbaak.info>
* contrib/scripts/install_prereq, /: Merged revisions 208542 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r208542 | mvanbaak | 2009-07-24 16:35:49 +0200 (Fri, 24
Jul 2009) | 13 lines use aptitude for debian based systems The
function to check wether we need to install packages was using
dpkg-query which was gives wrong output on Debian 5 Also, the
apt-get has been replaced with aptitude because aptitude is now
the preferred way to handle packages on Debian (closes issue
#15570) Reported by: mvanbaak Patches:
2009072400_installprereq-aptitude.diff uploaded by mvanbaak
(license 7) ........
2009-07-23 22:31 +0000 [r208501] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk/frame.h, main/rtp.c, main/channel.c,
main/udptl.c, main/frame.c, /, channels/chan_sip.c,
apps/app_fax.c, UPGRADE.txt, include/asterisk/udptl.h: Merged
revisions 208464 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r208464 |
kpfleming | 2009-07-23 16:57:24 -0500 (Thu, 23 Jul 2009) | 46
lines Rework of T.38 negotiation and UDPTL API to address
interoperability problems Over the past couple of months, a
number of issues with Asterisk negotiating (and successfully
completing) T.38 sessions with various endpoints have been found.
This patch attempts to address many of them, primarily focused
around ensuring that the endpoints' MaxDatagram size is honored,
and in addition by ensuring that T.38 session parameter
negotiation is performed correctly according to the ITU T.38
Recommendation. The major changes here are: 1) T.38 applications
in Asterisk (app_fax) only generate/receive IFP packets, they do
not ever work with UDPTL packets. As a result of this, they
cannot be allowed to generate packets that would overflow the
other endpoints' MaxDatagram size after the UDPTL stack adds any
error correction information. With this patch, the application is
told the maximum *IFP* size it can generate, based on a
calculation using the far end MaxDatagram size and the active
error correction mode on the T.38 session. The same is true for
sending *our* MaxDatagram size to the remote endpoint; it is
computed from the value that the application says it can accept
(for a single IFP packet) combined with the active error
correction mode. 2) All treatment of T.38 session parameters as
'capabilities' in chan_sip has been removed; these parameters are
not at all like audio/video stream capabilities. There are strict
rules to follow for computing an answer to a T.38 offer, and
chan_sip now follows those rules, using the desired parameters
from the application (or channel) that wants to accept the T.38
negotiation. 3) chan_sip now stores and forwards
ast_control_t38_parameters structures for tracking 'our' and
'their' T.38 session parameters; this greatly simplifies
negotiation, especially for pass-through calls. 4) Since T.38
negotiation without specifying parameters or receiving the final
negotiated parameters is not very worthwhile, the AST_CONTROL_T38
control frame has been removed. A note has been added to
UPGRADE.txt about this removal, since any out-of-tree
applications that use it will no longer function properly until
they are upgraded to use AST_CONTROL_T38_PARAMETERS. Review:
https://reviewboard.asterisk.org/r/310/ ........
2009-07-23 19:36 +0000 [r208391] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 208388 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r208388 | mmichelson | 2009-07-23 14:34:49 -0500 (Thu, 23 Jul
2009) | 24 lines Merged revisions 208386 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r208386 | mmichelson | 2009-07-23 14:24:21 -0500 (Thu, 23 Jul
2009) | 17 lines Fix a problem where a 491 response could be sent
out of dialog. This generalizes the fix for issue 13849. The
initial fix corrected the problem that Asterisk would reply with
a 491 if a reinvite were received from an endpoint and we had not
yet received an ACK from that endpoint for the initial INVITE it
had sent us. This expansion also allows Asterisk to appropriately
handle an INVITE with authorization credentials if Asterisk had
not received an ACK from the previous transaction in which
Asterisk had responded to an unauthorized INVITE with a 407.
(closes issue #14239) Reported by: klaus3000 Patches: 14239.patch
uploaded by mmichelson (license 60) Tested by: klaus3000 ........
................
2009-07-23 19:25 +0000 [r208387] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 208383 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r208383 | jpeeler | 2009-07-23 14:21:50 -0500
(Thu, 23 Jul 2009) | 12 lines Merged revisions 208380 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r208380 | jpeeler | 2009-07-23 14:19:53 -0500 (Thu, 23 Jul 2009)
| 6 lines Only set the priindication setting when not performing
a reload (closes issue #14696) Reported by: fdecher ........
................
2009-07-23 16:30 +0000 [r208266-208320] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 208314 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r208314 | mmichelson | 2009-07-23 11:29:37 -0500 (Thu, 23 Jul
2009) | 9 lines Merged revisions 208312 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r208312 | mmichelson | 2009-07-23 11:29:18 -0500 (Thu, 23 Jul
2009) | 3 lines Remove inaccurate XXX comment. ........
................
* /, channels/chan_sip.c: Merged revisions 208263 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r208263 | mmichelson | 2009-07-23 10:46:34 -0500 (Thu, 23 Jul
2009) | 15 lines Merged revisions 208262 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r208262 | mmichelson | 2009-07-23 10:43:07 -0500 (Thu, 23 Jul
2009) | 8 lines Properly handle 183 responses which do not
contain an SDP. (closes issue #15442) Reported by: ffloimair
Patches: 15442.patch uploaded by mmichelson (license 60) Tested
by: tkarl, ffloimair ........ ................
2009-07-22 21:46 +0000 [r208116] Jason Parker <jparker@digium.com>
* /, apps/app_festival.c: Merged revisions 208113 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r208113 |
qwell | 2009-07-22 16:43:57 -0500 (Wed, 22 Jul 2009) | 9 lines
Restore an int declaration on PPC platforms. This x is one crafty
little bugger... It was used for 2 different things (one of which
was only done on PPC) in 1.4. One of the uses were removed in
trunk, and with it went the declaration. (closes issue #14038)
Reported by: ffloimair ........
2009-07-22 16:52 +0000 [r207949-208053] Tilghman Lesher <tlesher@digium.com>
* /, res/res_realtime.c: Merged revisions 208052 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r208052 |
tilghman | 2009-07-22 11:49:42 -0500 (Wed, 22 Jul 2009) | 7 lines
Clarify documentation on 'realtime update2' to show more than one
condition. (closes issue #15357) Reported by: snuffy Patches:
bug_fix_doc_update2.diff uploaded by snuffy (license 35)
(slightly modified by me) ........
* /, funcs/func_strings.c: Merged revisions 207946 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r207946 | tilghman | 2009-07-21 17:45:32 -0500
(Tue, 21 Jul 2009) | 15 lines Merged revisions 207945 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r207945 | tilghman | 2009-07-21 17:38:54 -0500 (Tue, 21 Jul 2009)
| 8 lines Force an error if a blank is passed to QUOTE (because
the documentation states the argument is not optional). This
change makes URIENCODE and QUOTE behave similarly, since the
documentation states that the argument is not optional, for both.
(closes issue #15439) Reported by: pkempgen Patches:
20090706__issue15439.diff.txt uploaded by tilghman (license 14)
........ ................
2009-07-21 22:23 +0000 [r207930] Russell Bryant <russell@digium.com>
* doc/CODING-GUIDELINES, /: Merged revisions 207925 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r207925 | russell | 2009-07-21 17:22:18 -0500 (Tue, 21 Jul 2009)
| 4 lines Note that we use tabs instead of spaces for
indentation. I'm surprised this was never actually in here...
........
2009-07-21 20:30 +0000 [r207785-207862] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 207854 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r207854 | jpeeler | 2009-07-21 15:26:02 -0500
(Tue, 21 Jul 2009) | 16 lines Merged revisions 207827 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r207827 | jpeeler | 2009-07-21 15:16:55 -0500 (Tue, 21 Jul 2009)
| 9 lines Wait for wink before dialing when using E&M wink
signaling There was already code for other signaling types in
dahdi_handle_event to handle dialing if a dial operation dial
string was present. Simply add SIG_EMWINK to the list. (closes
issue #14434) Reported by: araasch ........ ................
* channels/chan_dahdi.c: Revert r207638, this approach could
potentially block for an unacceptable amount of time.
2009-07-21 14:32 +0000 [r207727] Mark Michelson <mmichelson@digium.com>
* main/manager.c, /: Merged revisions 207723 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r207723 | mmichelson | 2009-07-21 09:29:40 -0500 (Tue, 21 Jul
2009) | 11 lines Merged revisions 207714 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r207714 | mmichelson | 2009-07-21 09:26:00 -0500 (Tue, 21 Jul
2009) | 5 lines Document default timeout for AMI originations.
AST-224 ........ ................
2009-07-21 13:56 +0000 [r207685] Kevin P. Fleming <kpfleming@digium.com>
* channels/Makefile, doc/video_console.txt, Makefile, agi/Makefile,
codecs/Makefile, utils/Makefile, funcs/Makefile,
codecs/lpc10/Makefile, main/db1-ast/Makefile, /, main/Makefile,
codecs/gsm/Makefile, Makefile.moddir_rules, Makefile.rules,
pbx/Makefile, res/Makefile: Merged revisions 207680 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r207680 | kpfleming | 2009-07-21 08:28:04 -0500
(Tue, 21 Jul 2009) | 18 lines Merged revisions 207647 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r207647 | kpfleming | 2009-07-21 08:04:44 -0500 (Tue, 21 Jul
2009) | 12 lines Ensure that user-provided CFLAGS and LDFLAGS are
honored. This commit changes the build system so that
user-provided flags (in ASTCFLAGS and ASTLDFLAGS) are supplied to
the compiler/linker *after* all flags provided by the build
system itself, so that the user can effectively override the
build system's flags if desired. In addition, ASTCFLAGS and
ASTLDFLAGS can now be provided *either* in the environment before
running 'make', or as variable assignments on the 'make' command
line. As a result, the use of COPTS and LDOPTS is no longer
necessary, so they are no longer documented, but are still
supported so as not to break existing build systems that supply
them when building Asterisk. ........ ................
2009-07-21 04:51 +0000 [r207638] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c: Wait for wink before dialing when using
E&M wink signaling This patch adds a new dahdi_wait function to
specifically wait for the wink event. If the wink is not
eventually received the channel is hung up. (closes issue #14434)
Reported by: araasch Patches: emwinkmod uploaded by araasch
(license 693)
2009-07-20 22:14 +0000 [r207523] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 207424 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r207424 | mmichelson | 2009-07-20 14:48:12 -0500 (Mon, 20 Jul
2009) | 39 lines Merged revisions 207423 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r207423 | mmichelson | 2009-07-20 14:39:59 -0500 (Mon, 20 Jul
2009) | 33 lines Answer video SDP offers properly when
videosupport is not enabled. Copied from Review board: In issue
12434, the reporter describes a situation in which audio and
video is offered on the call, but because videosupport is
disabled in sip.conf, Asterisk gives no response at all to the
video offer. According to RFC 3264, all media offers should have
a corresponding answer. For offers we do not intend to actually
reply to with meaningful values, we should still reply with the
port for the media stream set to 0. In this patch, we take note
of what types of media have been offered and save the information
on the sip_pvt. The SDP in the response will take into account
whether media was offered. If we are not otherwise going to
answer a media offer, we will insert an appropriate m= line with
the port set to 0. It is important to note that this patch is
pretty much a bandage being applied to a broken bone. The patch
*only* helps for situations where video is offered but
videosupport is disabled and when udptl_pt is disabled but T.38
is offered. Asterisk is not guaranteed to respond to every media
offer. Notable cases are when multiple streams of the same type
are offered. The 2 media stream limit is still present with this
patch, too. In trunk and the 1.6.X branches, things will be a bit
different since Asterisk also supports text in SDPs as well.
(closes issue #12434) Reported by: mnnojd Review:
https://reviewboard.asterisk.org/r/311 Review:
https://reviewboard.asterisk.org/r/313 ........ ................
2009-07-20 16:41 +0000 [r207364] Russell Bryant <russell@digium.com>
* main/channel.c, /: Merged revisions 207361 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r207361 | russell | 2009-07-20 11:36:15 -0500 (Mon, 20 Jul 2009)
| 16 lines Merged revisions 207360 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r207360 | russell | 2009-07-20 11:26:24 -0500 (Mon, 20 Jul 2009)
| 9 lines Only do the chan->fdno check in ast_read() in a
developer build. I changed this check to only happen in a
dev-mode build. I also added a comment explaining what is going
on. I also made it so that detection of this situation does not
affect ast_read() operation. (closes issue #14723) Reported by:
seadweller ........ ................
2009-07-18 04:19 +0000 [r207327] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c, /: Merged revisions 207317 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r207317 | tilghman | 2009-07-17 23:16:44 -0500 (Fri, 17 Jul 2009)
| 3 lines Flag field in wrong position. Reported by "Hoggins!" on
asterisk-dev list. ........
2009-07-18 03:50 +0000 [r207315] Richard Mudgett <rmudgett@digium.com>
* channels/misdn/isdn_lib.c, channels/chan_misdn.c: Merged
revisions 145293,158010 from
https://origsvn.digium.com/svn/asterisk/branches/1.4 to make
merging easier. These changes are already on trunk.
................ r145293 | rmudgett | 2008-09-30 18:55:24 -0500
(Tue, 30 Sep 2008) | 54 lines channels/chan_misdn.c
channels/misdn/isdn_lib.c * Miscellaneous other fixes from trunk
to make merging easier later. ........ r145200 | rmudgett |
2008-09-30 16:00:54 -0500 (Tue, 30 Sep 2008) | 7 lines *
Miscellaneous formatting changes to make v1.4 and trunk more
merge compatible in the mISDN area. channels/chan_misdn.c *
Eliminated redundant code in cb_events() EVENT_SETUP ........
r144257 | crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008)
| 9 lines improved helptext of misdn_set_opt. ........ r142181 |
rmudgett | 2008-09-09 12:30:52 -0500 (Tue, 09 Sep 2008) | 1 line
Cleaned up comment ........ r138738 | rmudgett | 2008-08-18
16:07:28 -0500 (Mon, 18 Aug 2008) | 30 lines
channels/chan_misdn.c * Made bearer2str() use
allowed_bearers_array[] * Made use the causes.h defines instead
of hardcoded numbers. * Made use Asterisk presentation indicator
values if either of the mISDN presentation or screen options are
negative. * Updated the misdn_set_opt application option
descriptions. * Renamed the awkward Caller ID presentation
misdn_set_opt application option value not_screened to
restricted. Deprecated the not_screened option value.
channels/misdn/isdn_lib.c * Made use the causes.h defines instead
of hardcoded numbers. * Fixed some spelling errors and typos. *
Added all defined facility code strings to fac2str().
channels/misdn/isdn_lib.h * Added doxygen comments to struct
misdn_bchannel. channels/misdn/isdn_lib_intern.h * Added doxygen
comments to struct misdn_stack. channels/misdn_config.c
configs/misdn.conf.sample * Updated the mISDN presentation and
screen parameter descriptions. doc/misdn.txt (doc/tex/misdn.tex)
* Updated the misdn_set_opt application option descriptions. *
Fixed some spelling errors and typos. ................ r158010 |
rmudgett | 2008-11-19 19:46:09 -0600 (Wed, 19 Nov 2008) | 9 lines
Merged revision 157977 from
https://origsvn.digium.com/svn/asterisk/team/group/issue8824
........ Fixes JIRA ABE-1726 The dial extension could be empty if
you are using MISDN_KEYPAD to control ISDN provider features.
................
2009-07-17 22:31 +0000 [r207226-207257] Tilghman Lesher <tlesher@digium.com>
* /, doc/voicemail_odbc_postgresql.txt: Merged revisions 207255 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r207255 | tilghman | 2009-07-17 17:29:50 -0500 (Fri, 17
Jul 2009) | 2 lines Add flag here, too (as requested by jsmith)
........
* /, doc/tex/odbcstorage.tex, UPGRADE.txt: Merged revisions 207224
via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r207224 | tilghman | 2009-07-17 17:04:43 -0500 (Fri, 17
Jul 2009) | 2 lines Document the "flag" field in the
voicemessages table. ........
2009-07-17 19:40 +0000 [r207104-207159] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 207156 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r207156 | jpeeler | 2009-07-17 14:37:38 -0500
(Fri, 17 Jul 2009) | 14 lines Merged revisions 207155 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r207155 | jpeeler | 2009-07-17 14:36:19 -0500 (Fri, 17 Jul 2009)
| 7 lines Fix format specifier to print out an unsigned long
long. Yep, it's even ifdefed out code. But it made it to the RR
list... (closes issue #14726) Reported by: lmadsen ........
................
* configs/chan_dahdi.conf.sample, /: Merged revisions 207095 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r207095 | jpeeler | 2009-07-17 14:16:35 -0500 (Fri, 17
Jul 2009) | 2 lines Update some missing allowed options for
overlapdial ........
2009-07-17 17:52 +0000 [r206869-207030] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 207029 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r207029 |
dvossel | 2009-07-17 12:51:44 -0500 (Fri, 17 Jul 2009) | 6 lines
sip option flags handled incorrectly (closes issue #15376)
Reported by: Takehiko Ooshima Tested by: dvossel,
Takehiko_Ooshima ........
* /, channels/chan_sip.c: Merged revisions 206939 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r206939 | dvossel | 2009-07-17 11:13:22 -0500 (Fri, 17 Jul 2009)
| 20 lines Merged revisions 206938 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r206938 | dvossel | 2009-07-17 11:05:06 -0500 (Fri, 17 Jul 2009)
| 14 lines SIP incorrect From: header information when callpres
is prohib Some ITSP make use of the "Anonymous" display name to
detect a requirement to withhold caller id across the PSTN. This
does not work if the display name is "Unknown". (closes issue
#14465) Reported by: Nick_Lewis Patches:
chan_sip.c-callerpres.patch uploaded by Nick (license 657)
chan_sip.c-callerpres_trunk.patch uploaded by dvossel (license
671) Tested by: Nick_Lewis, dvossel ........ ................
* /, funcs/func_timeout.c: Merged revisions 206877 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r206877 | dvossel | 2009-07-16 16:45:14 -0500 (Thu, 16 Jul 2009)
| 6 lines TIMEOUT(absolute) returned negative value. (closes
issue #15513) Reported by: ys ........
* configs/iax.conf.sample, /: Merged revisions 206873 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r206873 | dvossel | 2009-07-16 16:33:51 -0500
(Thu, 16 Jul 2009) | 12 lines Merged revisions 206872 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r206872 | dvossel | 2009-07-16 16:33:19 -0500 (Thu, 16 Jul 2009)
| 6 lines error in iax.conf related IP-based access control
(closes issue #15518) Reported by: pkempgen ........
................
* /, main/callerid.c: Merged revisions 206868 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r206868 | dvossel | 2009-07-16 16:25:22 -0500 (Thu, 16 Jul 2009)
| 14 lines Merged revisions 206867 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r206867 | dvossel | 2009-07-16 16:24:16 -0500 (Thu, 16 Jul 2009)
| 8 lines avoid segfault caused by user error If the CALLERPRES()
dialplan function is set to nothing, a segfault occurs. This is
user error to begin with, but I'd rather see a cli warning
message than have Asterisk crash on me. ........ ................
2009-07-16 16:53 +0000 [r206811] Tilghman Lesher <tlesher@digium.com>
* funcs/func_realtime.c, /: Merged revisions 206808 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r206808 | tilghman | 2009-07-16 11:51:05 -0500
(Thu, 16 Jul 2009) | 13 lines Merged revisions 206807 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r206807 | tilghman | 2009-07-16 11:27:35 -0500 (Thu, 16 Jul 2009)
| 6 lines Fix a memory leak. (closes issue #15517) Reported by:
adomjan Patches: func_realtime.c-ast_variable_destroy.diff
uploaded by adomjan (license 487) ........ ................
2009-07-15 22:04 +0000 [r206770] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 206768 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r206768 |
dvossel | 2009-07-15 17:04:13 -0500 (Wed, 15 Jul 2009) | 8 lines
Session timer were not activated if Supported header field in
INVITE had both "timer" and other options. (closes issue #15403)
Reported by: makoto Patches: sip-session-timer.patch uploaded by
makoto (license ........
2009-07-15 21:50 +0000 [r206765] Richard Mudgett <rmudgett@digium.com>
* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h, /:
Merged revisions 206707 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r206707 | rmudgett | 2009-07-15 16:14:41 -0500 (Wed, 15 Jul 2009)
| 33 lines Merged revisions 206706 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
................ r206706 | rmudgett | 2009-07-15 15:44:55 -0500
(Wed, 15 Jul 2009) | 26 lines Merged revision 206700 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.2-...
.......... Fixed chan_misdn crash because mISDNuser library is
not thread safe. With Asterisk the mISDNuser library is driven by
two threads concurrently: 1.
channels/misdn/isdn_lib.c::manager_event_handler() 2.
channels/misdn/isdn_lib.c::misdn_lib_isdn_event_catcher() Calls
into the library are done concurrently and recursively from
isdn_lib.c. Both threads can fiddle with the master/child
layer3_proc_t lists. One thread may traverse the list when the
other interrupts it and then removes the list element which the
first thread was currently handling. This is exactly what caused
the crash. About 60 calls were needed to a Gigaset CX475 before
it occurred once. This patch adds locking when calling into the
mISDNuser library. This also fixes some cb_log calls with wrong
port parameter. JIRA ABE-1913 Patches: misdn-locking.patch
(Modified with mostly cosmetic changes) ..........
................ ................
2009-07-15 20:20 +0000 [r206703] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 206702 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r206702 |
dvossel | 2009-07-15 15:20:01 -0500 (Wed, 15 Jul 2009) | 10 lines
callerid(num) is wrong when username is missing A domain only sip
uri <sip:123.123.123.123> would return 123.123.123.123 as callid
num. Now, if the username is missing from a uri, the callerid num
field is left empty. (closes issue #15476) Reported by: viraptor
........
2009-07-15 16:04 +0000 [r206639] Sean Bright <sean@malleable.com>
* codecs/codec_dahdi.c, /: Merged revisions 206636 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r206636 | seanbright | 2009-07-15 12:00:24 -0400
(Wed, 15 Jul 2009) | 9 lines Merged revisions 206635 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r206635 | seanbright | 2009-07-15 11:57:51 -0400 (Wed,
15 Jul 2009) | 1 line Only print debug info in codec_dahdi if we
are asking for it. ........ ................
2009-07-14 20:26 +0000 [r206598] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_meetme.c, contrib/scripts/meetme.sql: Merged
revisions 206567 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r206567 |
tilghman | 2009-07-14 15:14:45 -0500 (Tue, 14 Jul 2009) | 6 lines
Document all meetme realtime fields, and in the process, make
some field lengths more consistent. (closes issue #15493)
Reported by: lasko Patches: meetme.diff uploaded by lasko
(license 833) ........
2009-07-14 19:49 +0000 [r206565] Richard Mudgett <rmudgett@digium.com>
* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
channels/chan_misdn.c, /: Merged revisions 206489 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r206489 | rmudgett | 2009-07-14 12:01:48 -0500
(Tue, 14 Jul 2009) | 35 lines Merged revisions 206487 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r206487 | rmudgett | 2009-07-14 11:44:47 -0500 (Tue, 14 Jul 2009)
| 28 lines Fixes several call transfer issues with chan_misdn. *
issue #14355 - Crash if attempt to transfer a call to an
application. Masquerade the other pair of the four asterisk
channels involved in the two calls. The held call already must be
a bridged call (not an applicaton) or it would have been
rejected. * issue #14692 - Held calls are not automatically
cleared after transfer. Allow the core to initate disconnect of
held calls to the ISDN port. This also fixes a similar case where
the party on hold hangs up before being transferred or taken off
hold. * JIRA ABE-1903 - Orphaned held calls left in
music-on-hold. Do not simply block passing the hangup event on
held calls to asterisk core. * Fixed to allow held calls to be
transferred to ringing calls. Previously, held calls could only
be transferred to connected calls. * Eliminated unused call
states to simplify hangup code. * Eliminated most uses of
"holded" because it is not a word. (closes issue #14355) (closes
issue #14692) Reported by: sodom Patches:
misdn_xfer_v14_r205839.patch uploaded by rmudgett (license 664)
Tested by: rmudgett ........ ................
2009-07-14 14:59 +0000 [r206389] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c, /: Merged revisions 206386 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r206386 | russell | 2009-07-14 09:51:44 -0500
(Tue, 14 Jul 2009) | 20 lines Merged revisions 206385 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
................ r206385 | russell | 2009-07-14 09:48:00 -0500
(Tue, 14 Jul 2009) | 13 lines Merged revisions 206384 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r206384 | russell | 2009-07-14 09:45:47 -0500 (Tue, 14 Jul 2009)
| 6 lines Ensure apathetic replies are sent out on the proper
socket. chan_iax2 supports multiple address bindings. The
send_apathetic_reply() function did not attempt to send its
response on the same socket that the incoming message came in on.
........ ................ ................
2009-07-14 01:59 +0000 [r206373] Richard Mudgett <rmudgett@digium.com>
* channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
revisions 206341 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r206341 | rmudgett | 2009-07-13 19:48:59 -0500 (Mon, 13 Jul 2009)
| 11 lines Merged revisions 206284 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r206284 | rmudgett | 2009-07-13 19:17:28 -0500 (Mon, 13 Jul 2009)
| 4 lines Fix some memory leaks in chan_misdn. JIRA ABE-1911
........ ................
2009-07-13 23:27 +0000 [r206281] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 206280 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r206280 |
dvossel | 2009-07-13 18:26:51 -0500 (Mon, 13 Jul 2009) | 9 lines
dns lookup of peername rather than peer's host in
transmit_register() (closes issue #15052) Reported by: fsantulli
Patches: chan_sip_bug_15052_[20090626204511].patch uploaded by
fsantulli (license 818) Tested by: fsantulli ........
2009-07-13 16:24 +0000 [r206187] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c, /: Merged revisions 206185 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r206185 | tilghman | 2009-07-13 11:23:07 -0500 (Mon, 13 Jul 2009)
| 2 lines Remove reference to non-existent help file ........
2009-07-10 21:46 +0000 [r205986] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 205985 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r205985 |
dvossel | 2009-07-10 16:42:10 -0500 (Fri, 10 Jul 2009) | 16 lines
SIP register not using peer's outbound proxy If callbackextension
is defined for a peer it successfully causes a registration to
occur, but the registration ignores the outboundproxy settings
for the peer. This patch allows the peer to be passed to
obproxy_get() in transmit_register(). (closes issue #14344)
Reported by: Nick_Lewis Patches:
callbackextension_peer_trunk.diff uploaded by dvossel (license
671) Tested by: dvossel Review:
https://reviewboard.asterisk.org/r/294/ ........
2009-07-10 18:45 +0000 [r205942] Kevin P. Fleming <kpfleming@digium.com>
* main/udptl.c, /: Merged revisions 205939 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r205939 |
kpfleming | 2009-07-10 13:44:09 -0500 (Fri, 10 Jul 2009) | 1 line
Update comments about the level of T.38 support in Asterisk.
........
2009-07-10 17:54 +0000 [r205882] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 205878 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r205878 | mmichelson | 2009-07-10 12:39:57 -0500 (Fri, 10 Jul
2009) | 30 lines Merged revisions 205877 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
................ r205877 | mmichelson | 2009-07-10 12:39:13 -0500
(Fri, 10 Jul 2009) | 23 lines Merged revisions 205776 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r205776 | mmichelson | 2009-07-10 10:56:45 -0500
(Fri, 10 Jul 2009) | 16 lines Merged revisions 205775 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul
2009) | 10 lines Ensure that outbound NOTIFY requests are
properly routed through stateful proxies. With this change, we
make note of Record-Route headers present in any SUBSCRIBE
request that we receive so that our outbound NOTIFY requests will
have the proper Route headers in them. (closes issue #14725)
Reported by: ibc ........ ................ ................
................
2009-07-10 16:47 +0000 [r205841] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 205840 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r205840 | dvossel | 2009-07-10 11:42:04 -0500 (Fri, 10 Jul 2009)
| 37 lines Merged revisions 205804 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r205804 | dvossel | 2009-07-10 11:23:59 -0500 (Fri, 10 Jul 2009)
| 31 lines SIP registration auth loop caused by stale nonce If an
endpoint sends two registration requests in a very short period
of time with the same nonce, both receive 401 responses from
Asterisk, each with a different nonce (the second 401 containing
the current nonce and the first one being stale). If the endpoint
responds to the first 401, it does not match the current nonce so
Asterisk sends a third 401 with a newly generated nonce (which
updates the current nonce)... Now if the endpoint responds to the
second 401, it does not match the current nonce either and
Asterisk sends a fourth 401 with a newly generated nonce... This
loop goes on and on. There appears to be a simple fix for this.
If the nonce from the request does not match our nonce, but is a
good response to a previous nonce, instead of sending a 401 with
a newly generated nonce, use the current one instead. This breaks
the loop as the nonce is not updated until a response is
received. Additional logic has been added to make sure no nonce
can be responded to twice though. (closes issue #15102) Reported
by: Jamuel Patches: patch-bug_0015102 uploaded by Jamuel (license
809) nonce_sip.diff uploaded by dvossel (license 671) Tested by:
Jamuel Review: https://reviewboard.asterisk.org/r/289/ ........
................
2009-07-10 16:01 +0000 [r205781] Kevin P. Fleming <kpfleming@digium.com>
* /, apps/app_fax.c: Merged revisions 205780 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r205780 |
kpfleming | 2009-07-10 11:00:44 -0500 (Fri, 10 Jul 2009) | 11
lines Eliminate extraneous LOG_DEBUG messages generated by
app_fax. The transmit_audio() and transmit_t38() functions in
app_fax have processing loops that are supposed to wait for
frames to arrive on the channel and then handle them, but they
also have short timeouts so that the loops can have watchdog
timers and do other required processing. This commit changes the
loops to not actually call ast_read() and attempt to process the
returned frame unless a frame actually arrived, eliminating
hundreds of LOG_DEBUG messages and slightly improving
performance. ........
2009-07-10 15:58 +0000 [r205779] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 205776 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r205776 | mmichelson | 2009-07-10 10:56:45 -0500 (Fri, 10 Jul
2009) | 16 lines Merged revisions 205775 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul
2009) | 10 lines Ensure that outbound NOTIFY requests are
properly routed through stateful proxies. With this change, we
make note of Record-Route headers present in any SUBSCRIBE
request that we receive so that our outbound NOTIFY requests will
have the proper Route headers in them. (closes issue #14725)
Reported by: ibc ........ ................
2009-07-10 15:36 +0000 [r205773] Kevin P. Fleming <kpfleming@digium.com>
* /, apps/app_fax.c: Merged revisions 205770 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r205770 |
kpfleming | 2009-07-10 10:28:11 -0500 (Fri, 10 Jul 2009) | 12
lines Fix some remaining T.38 negotiation problems in app_fax.
Revision 205696 did not quite fix all the issues with the T.38
negotiation changes and app_fax; this patch corrects them, along
with a couple of other minor issues. (closes issue #15480)
Reported by: dimas Patches: test2-15480.patch uploaded by dimas
(license 88) ........
2009-07-09 23:56 +0000 [r205731] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: Merged revisions 205728 via svn merge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r205728 | rmudgett | 2009-07-09 18:37:53 -0500 (Thu, 09 Jul 2009)
| 21 lines No audio on calls from Asterisk to various ISDN
devices until DTMF sent by caller. Add missing clearing of the
dialing flag when the ISDN call is CONNECTED. (i.e. When libpri
generates the event PRI_EVENT_ANSWER.) (closes issue #15420)
Reported by: scottbmilne Patches: bug15420-1.4.25.1-diff2.txt
uploaded by alecdavis (license 585) Tested by: scottbmilne,
alecdavis (closes issue #15416) Reported by: avinoash (closes
issue #15389) Reported by: alecdavis This patch should also fix
the following issue: (issue #15205) Reported by: vinsik ........
2009-07-09 21:27 +0000 [r205699] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk/frame.h, /, channels/chan_sip.c, apps/app_fax.c:
Merged revisions 205696 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r205696 |
kpfleming | 2009-07-09 16:20:23 -0500 (Thu, 09 Jul 2009) | 16
lines Repair ability of SendFAX/ReceiveFAX to respond to T.38
switchover. Recent changes in T.38 negotiation in Asterisk caused
these applications to not respond when the other endpoint
initiated a switchover to T.38; this resulted in the T.38
switchover failing, and the FAX attempt to be made using an audio
connection, instead of T.38 (which would usually cause the FAX to
fail completely). This patch corrects this problem, and the
applications will now correctly respond to the T.38 switchover
request. In addition, the response will include the appopriate
T.38 session parameters based on what the other end offered and
what our end is capable of. (closes issue #14849) Reported by:
afosorio ........
2009-07-09 16:19 +0000 [r205595-205603] David Vossel <dvossel@digium.com>
* include/asterisk/time.h, /: Merged revisions 205600 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r205600 | dvossel | 2009-07-09 11:19:09 -0500
(Thu, 09 Jul 2009) | 9 lines Merged revisions 205599 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r205599 | dvossel | 2009-07-09 11:18:09 -0500 (Thu, 09
Jul 2009) | 2 lines Changing ast_samp2tv to not use floating
point. ........ ................
* channels/chan_iax2.c, include/asterisk/frame.h, main/rtp.c, /:
Merged revisions 205479 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r205479 | dvossel | 2009-07-08 18:19:09 -0500 (Wed, 08 Jul 2009)
| 16 lines Merged revisions 205471 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r205471 | dvossel | 2009-07-08 18:15:54 -0500 (Wed, 08 Jul 2009)
| 10 lines Fixes 8khz assumptions Many calculations assume 8khz
is the codec rate. This is not always the case. This patch only
addresses chan_iax.c and res_rtp_asterisk.c, but I am sure there
are other areas that make this assumption as well. Review:
https://reviewboard.asterisk.org/r/306/ ........ ................
2009-07-09 08:34 +0000 [r205535] Michiel van Baak <michiel@vanbaak.info>
* /, main/ssl.c: Merged revisions 205532 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r205532 |
mvanbaak | 2009-07-09 10:31:24 +0200 (Thu, 09 Jul 2009) | 5 lines
pthread_self returns a pthread_t which is not an unsigned int on
all pthread implementations. Casting it to an unsigned int fixes
compiler warnings. Tested on OpenBSD and Linux both 32 and 64 bit
........
2009-07-08 22:15 +0000 [r205411-205413] David Vossel <dvossel@digium.com>
* include/asterisk/pbx.h, include/asterisk/devicestate.h,
main/pbx.c, /, main/devicestate.c: Merged revisions 205412 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r205412 | dvossel | 2009-07-08 17:15:06 -0500
(Wed, 08 Jul 2009) | 12 lines Merged revisions 205409 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r205409 | dvossel | 2009-07-08 16:35:12 -0500 (Wed, 08 Jul 2009)
| 6 lines moving ast_devstate_to_extenstate to pbx.c from
devicestate.c ast_devstate_to_extenstate belongs in pbx.c. This
change fixes a compile time error with chan_vpb as well. ........
................
* /, main/devicestate.c: Merged revisions 205410 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r205410 |
dvossel | 2009-07-08 17:02:54 -0500 (Wed, 08 Jul 2009) | 3 lines
missing comma in devstatestring array ........
2009-07-08 19:28 +0000 [r205353] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c, /: Merged revisions 205350 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r205350 | mmichelson | 2009-07-08 14:26:55 -0500 (Wed, 08 Jul
2009) | 20 lines Merged revisions 205349 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r205349 | mmichelson | 2009-07-08 14:26:13 -0500 (Wed, 08 Jul
2009) | 14 lines Prevent phantom calls to queue members. If a
caller were to hang up while a periodic announcement or position
were being said, the return value for those functions would
incorrectly indicate that the caller was still in the queue. With
these changes, the problem does not occur. (closes issue #14631)
Reported by: latinsud Patches: queue_announce_ghost_call2.diff
uploaded by latinsud (license 745) (with small modification from
me) ........ ................
2009-07-08 18:22 +0000 [r205302] Jason Parker <jparker@digium.com>
* config.guess, config.sub, /: Merged revisions 205291 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r205291 | qwell | 2009-07-08 13:19:46 -0500
(Wed, 08 Jul 2009) | 9 lines Merged revisions 205288 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r205288 | qwell | 2009-07-08 13:19:03 -0500 (Wed, 08 Jul
2009) | 1 line Update config.guess and config.sub from the
savannah.gnu.org git repo. ........ ................
2009-07-08 18:18 +0000 [r205287] David Brooks <dbrooks@digium.com>
* /, main/features.c: Merged revisions 205254 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r205254 |
dbrooks | 2009-07-08 12:26:26 -0500 (Wed, 08 Jul 2009) | 8 lines
Fixes Park() argument handling Park() was not respecting the
arguments passed to it. Any extension/context/priority given to
it was being ignored. This patch remedies this. (closes issue
#15380) Reported by: DLNoah ........
2009-07-08 17:00 +0000 [r205223] Tilghman Lesher <tlesher@digium.com>
* main/say.c: oops, fixing build
2009-07-08 16:55 +0000 [r205217] David Vossel <dvossel@digium.com>
* include/asterisk/time.h, /: Merged revisions 205216 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r205216 | dvossel | 2009-07-08 11:54:24 -0500
(Wed, 08 Jul 2009) | 17 lines Merged revisions 205215 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r205215 | dvossel | 2009-07-08 11:53:40 -0500 (Wed, 08 Jul 2009)
| 10 lines ast_samp2tv needs floating point for 16khz audio In
ast_samp2tv(), (1000000 / _rate) = 62.5 when _rate is 16000. The
.5 is currently stripped off because we don't calculate using
floating points. This causes madness with 16khz audio. (issue
ABE-1899) Review: https://reviewboard.asterisk.org/r/305/
........ ................
2009-07-08 16:30 +0000 [r205207] Tilghman Lesher <tlesher@digium.com>
* /, main/say.c: Merged revisions 205196 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r205196 | tilghman | 2009-07-08 11:27:50 -0500 (Wed, 08 Jul 2009)
| 9 lines Merged revisions 205188 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r205188 | tilghman | 2009-07-08 11:26:15 -0500 (Wed, 08 Jul 2009)
| 2 lines Add redirection warnings for the invalid language codes
previously removed. ........ ................
2009-07-08 15:57 +0000 [r205148-205154] Russell Bryant <russell@digium.com>
* /, main/ssl.c: Merged revisions 205151 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r205151 |
russell | 2009-07-08 10:56:28 -0500 (Wed, 08 Jul 2009) | 2 lines
Use tabs instead of spaces for indentation. ........
* include/asterisk/_private.h, res/res_jabber.c, main/asterisk.c,
/, main/Makefile, res/res_crypto.c, main/ssl.c (added): Merged
revisions 205120 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r205120 |
russell | 2009-07-08 10:17:19 -0500 (Wed, 08 Jul 2009) | 16 lines
Move OpenSSL initialization to a single place, make library usage
thread-safe. While doing some reading about OpenSSL, I noticed a
couple of things that needed to be improved with our usage of
OpenSSL. 1) We had initialization of the library done in multiple
modules. This has now been moved to a core function that gets
executed during Asterisk startup. We already link OpenSSL into
the core for TCP/TLS functionality, so this was the most logical
place to do it. 2) OpenSSL is not thread-safe by default.
However, making it thread safe is very easy. We just have to
provide a couple of callbacks. One callback returns a thread ID.
The other handles locking. For more information, start with the
"Is OpenSSL thread-safe?" question on the FAQ page of
openssl.org. ........
2009-07-06 13:41 +0000 [r204951] Kevin P. Fleming <kpfleming@digium.com>
* main/channel.c, /: Merged revisions 204948 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r204948 |
kpfleming | 2009-07-06 08:38:29 -0500 (Mon, 06 Jul 2009) | 7
lines Improve handling of AST_CONTROL_T38 and
AST_CONTROL_T38_PARAMETERS for non-T.38-capable channels. This
change allows applications that request T.38 negotiation on a
channel that does not support it to get the proper indication
that it is not supported, rather than thinking that negotiation
was started when it was not. ........
2009-07-02 22:06 +0000 [r204838] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c, /: Merged revisions 204835 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r204835 | rmudgett | 2009-07-02 17:01:28 -0500
(Thu, 02 Jul 2009) | 17 lines Merged revisions 204834 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r204834 | rmudgett | 2009-07-02 16:59:43 -0500 (Thu, 02 Jul 2009)
| 10 lines Removed confusing warning message "Got Busy in
Connected State" If an incoming mISDN call is answered with the
Answer application and a subsequent Dial gets a busy endpoint
then it is valid for that already connected channel to get the
busy indication. Asterisk will play the busy tones until the
dialplan plays something else or hangs up the call. (closes issue
#11974) Reported by: fvdb ........ ................
2009-07-02 16:12 +0000 [r204711] David Vossel <dvossel@digium.com>
* include/asterisk/devicestate.h, main/pbx.c, /,
main/devicestate.c: Merged revisions 204710 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r204710 | dvossel | 2009-07-02 11:03:44 -0500 (Thu, 02 Jul 2009)
| 21 lines Merged revisions 204681 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r204681 | dvossel | 2009-07-02 10:05:57 -0500 (Thu, 02 Jul 2009)
| 14 lines Improved mapping of extension states from combined
device states. This fixes a few issues with incorrect extension
states and adds a cli command, core show device2extenstate, to
display all possible state mappings. (closes issue #15413)
Reported by: legart Patches: exten_helper.diff uploaded by
dvossel (license 671) Tested by: dvossel, legart, amilcar Review:
https://reviewboard.asterisk.org/r/301/ ........ ................
2009-06-30 21:30 +0000 [r204611] Tilghman Lesher <tlesher@digium.com>
* /, main/say.c, UPGRADE.txt: Merged revisions 204563 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r204563 | tilghman | 2009-06-30 15:41:04 -0500
(Tue, 30 Jun 2009) | 13 lines Merged revisions 204556 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r204556 | tilghman | 2009-06-30 15:23:51 -0500 (Tue, 30 Jun 2009)
| 6 lines More incorrect language codes, plus ensuring that
regionalizations use the specified language, and not English for
grammar. (closes issue #15022) Reported by: greenfieldtech
Patches: 20090519__issue15022.diff.txt uploaded by tilghman
(license 14) ........ ................
2009-06-30 18:55 +0000 [r204478] Jason Parker <jparker@digium.com>
* /, main/say.c: Merged revisions 204475 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r204475 | qwell | 2009-06-30 13:48:35 -0500 (Tue, 30 Jun 2009) |
9 lines Merged revisions 204474 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r204474 | qwell | 2009-06-30 13:47:06 -0500 (Tue, 30 Jun 2009) |
1 line Fix ast_say_counted_noun to correctly handle Polish. Fix a
comment typo in passing. ........ ................
2009-06-30 18:44 +0000 [r204473] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c, /, main/say.c, UPGRADE.txt: Recorded merge
of revisions 204470 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r204470 | tilghman | 2009-06-30 13:36:24 -0500 (Tue, 30 Jun 2009)
| 18 lines Recorded merge of revisions 204469 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r204469 | tilghman | 2009-06-30 13:23:35 -0500 (Tue, 30 Jun 2009)
| 11 lines "tw" is the language specification for Twi (from
Ghana) not Taiwanese. (closes issue #15346) Reported by: volivier
Patches: 20090617__issue15346__1.4.diff.txt uploaded by tilghman
(license 14) 20090617__issue15346__trunk.diff.txt uploaded by
tilghman (license 14) 20090617__issue15346__1.6.0.diff.txt
uploaded by tilghman (license 14)
20090617__issue15346__1.6.1.diff.txt uploaded by tilghman
(license 14) 20090617__issue15346__1.6.2.diff.txt uploaded by
tilghman (license 14) Tested by: volivier ........
................
2009-06-30 17:22 +0000 [r204442] Russell Bryant <russell@digium.com>
* configs/res_config_sqlite.conf (removed),
configs/res_config_sqlite.conf.sample (added), /: Merged
revisions 204440 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r204440 |
russell | 2009-06-30 12:22:16 -0500 (Tue, 30 Jun 2009) | 2 lines
Rename res_config_sqlite.conf to res_config_sqlite.conf.sample
(missing .sample). ........
2009-06-29 22:53 +0000 [r204250-204304] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 204301 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r204301 | mmichelson | 2009-06-29 17:50:35 -0500 (Mon, 29 Jun
2009) | 15 lines Merged revisions 204300 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r204300 | mmichelson | 2009-06-29 17:45:34 -0500 (Mon, 29 Jun
2009) | 9 lines Add error message so that it is clear why a SIP
peer was not processed when a DNS lookup fails on a host or
outboundproxy. (closes issue #13432) Reported by: p_lindheimer
Patches: outboundproxy.patch uploaded by p (license 558) ........
................
* /, channels/chan_sip.c: Merged revisions 204247 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r204247 | mmichelson | 2009-06-29 16:48:54 -0500 (Mon, 29 Jun
2009) | 32 lines Merged revisions 204243,204246 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r204243 | mmichelson | 2009-06-29 16:23:43 -0500 (Mon, 29 Jun
2009) | 22 lines Fix a problem where chan_sip would ignore "old"
but valid responses. chan_sip has had a problem for quite a long
time that would manifest when Asterisk would send multiple SIP
responses on the same dialog before receiving a response. The
problem occurred because chan_sip only kept track of the highest
outgoing sequence number used on the dialog. If Asterisk sent two
requests out, and a response arrived for the first request sent,
then Asterisk would ignore the response. The result was that
Asterisk would continue retransmitting the requests and ignoring
the responses until the maximum number of retransmissions had
been reached. The fix here is to rearrange the code a bit so that
instead of simply comparing the sequence number of the response
to our latest outgoing sequence number, we walk our list of
outstanding packets and determine if there is a match. If there
is, we continue. If not, then we ignore the response. In doing
this, I found a few completely useless variables that I have now
removed. (closes issue #11231) Reported by: flefoll Review:
https://reviewboard.asterisk.org/r/298 ........ r204246 |
mmichelson | 2009-06-29 16:37:05 -0500 (Mon, 29 Jun 2009) | 3
lines Fix build oops. ........ ................
2009-06-27 09:55 +0000 [r203961] Russell Bryant <russell@digium.com>
* CHANGES, /: Merged revisions 203960 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r203960 |
russell | 2009-06-27 04:51:45 -0500 (Sat, 27 Jun 2009) | 2 lines
Minor tweaks and spelling fixes for CHANGES and UPGRADE.txt.
........
2009-06-27 01:24 +0000 [r203941] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 203909 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r203909 | rmudgett | 2009-06-26 20:07:52 -0500
(Fri, 26 Jun 2009) | 23 lines Merged revisions 203908 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r203908 | rmudgett | 2009-06-26 19:55:12 -0500 (Fri, 26 Jun 2009)
| 16 lines The ISDN CPE side should not exclusively pick B
channels normally. Before this patch, Asterisk unconditionally
picked B channels exclusively on the CPE side and normally
allowed alternative B channels on the network side. Now Asterisk
does the opposite. Reasons for the CPE side to normally not pick
B channels exclusively: * For CPE point-to-multipoint mode (i.e.
phone side), the CPE side does not have enough information to
exclusively pick B channels. (There may be other devices on the
line.) * Q.931 gives preference to the network side picking B
channels. * Some telcos require the CPE side to not pick B
channels exclusively. (closes issue #14383) Reported by:
mbrancaleoni ........ ................
2009-06-26 22:14 +0000 [r203857] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 203853 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r203853 | jpeeler | 2009-06-26 17:11:31 -0500
(Fri, 26 Jun 2009) | 12 lines Merged revisions 203848 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r203848 | jpeeler | 2009-06-26 17:09:19 -0500 (Fri, 26 Jun 2009)
| 5 lines Make sure to recreate the dahdi pseudo channel after
dahdi restart (closes issue #14477) Reported by: timking ........
................
2009-06-26 21:27 +0000 [r203782-203828] Russell Bryant <russell@digium.com>
* /, main/file.c: Merged revisions 203802 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r203802 | russell | 2009-06-26 16:21:48 -0500 (Fri, 26 Jun 2009)
| 22 lines Merged revisions 203785 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r203785 | russell | 2009-06-26 16:16:39 -0500 (Fri, 26 Jun 2009)
| 15 lines Don't fast forward past the end of a message. This is
nice change for users of the voicemail application. If someone
gets a little carried away with fast forwarding through a
message, they can easily get to the end and accidentally exit the
voicemail application by hitting the fast forward key during the
following prompt. This adds some safety by not allowing a fast
forward past the end of a message. (closes issue #14554) Reported
by: lacoursj Patches: 21761.patch uploaded by lacoursj (license
707) Tested by: lacoursj ........ ................
* /, channels/chan_sip.c: Merged revisions 203779 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r203779 |
russell | 2009-06-26 15:45:00 -0500 (Fri, 26 Jun 2009) | 5 lines
Ensure the TCP read buffer is fully initialized before handling
each packet. (closes issue #14452) Reported by: umberto71
........
2009-06-26 20:18 +0000 [r203731] David Brooks <dbrooks@digium.com>
* apps/app_voicemail.c, /: Merged revisions 203721 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r203721 | dbrooks | 2009-06-26 15:13:51 -0500 (Fri, 26 Jun 2009)
| 16 lines Fixing voicemail's error in checking max silence vs
min message length Max silence was represented in milliseconds,
yet vmminsecs (minmessage) was represented as seconds. Also, the
inequality was reversed. The warning, if triggered, was "Max
silence should be less than minmessage or you may get empty
messages", which should have been logged if max silence was
greater than minmessage, but the check was for less than. Also,
conforming if statement to coding guidelines. closes issue
#15331) Reported by: markd Review:
https://reviewboard.asterisk.org/r/293/ ........
2009-06-26 19:49 +0000 [r203715] Russell Bryant <russell@digium.com>
* include/asterisk/devicestate.h, main/pbx.c, /,
main/devicestate.c: Merged revisions 203702 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r203702 |
russell | 2009-06-26 14:31:14 -0500 (Fri, 26 Jun 2009) | 5 lines
Make invalid hints report Unavailable instead of Idle. (closes
issue #14413) Reported by: pj ........
2009-06-26 19:48 +0000 [r203712] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c, /: Merged revisions 203710 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r203710 | dvossel | 2009-06-26 14:47:11 -0500 (Fri, 26 Jun 2009)
| 7 lines moving debug message from level 0 to 1. (closes issue
#15404) Reported by: leobrown Patches: iax_codec_debug.patch
uploaded by leobrown (license 541) ........
2009-06-26 19:42 +0000 [r203709] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 203672 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r203672 | jpeeler | 2009-06-26 14:03:25 -0500 (Fri, 26 Jun 2009)
| 16 lines Check if polarityonanswerdelay has elapsed before
setting a channel as answered after a polarity reversal.
Previously on a polarity switch event chan_dahdi would set the
channel immediately as answered. This would cause problems if a
polarity reversal occurred when the line was picked up as the
dial would not have yet occurred. Now if the polarity reversal
occurs before delay has elapsed after coming off hook or an
answer, it is ignored. Also, some refactoring was done in
_handle_event. (closes issue #13917) Reported by: alecdavis
Patches: chan_dahdi.bug13917.feb09.diff2.txt uploaded by
alecdavis (license 585) Tested by: alecdavis ........
2009-06-26 19:38 +0000 [r203705] Joshua Colp <jcolp@digium.com>
* configs/sip.conf.sample, include/asterisk/frame.h, main/rtp.c,
main/channel.c, main/frame.c, /, channels/chan_sip.c,
apps/app_fax.c: Merged revisions 203699 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r203699 |
file | 2009-06-26 16:27:24 -0300 (Fri, 26 Jun 2009) | 2 lines
Improve T.38 negotiation by exchanging session parameters between
application and channel. ........
2009-06-25 21:46 +0000 [r203445] David Vossel <dvossel@digium.com>
* main/ast_expr2.fl, main/ast_expr2.c, /: Merged revisions 203444
via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r203444 | dvossel | 2009-06-25 16:45:32 -0500 (Thu, 25
Jun 2009) | 4 lines fixes a few redundant conditions (issue
#15269) ........
2009-06-25 21:21 +0000 [r203400] Terry Wilson <twilson@digium.com>
* main/cli.c, /: Merged revisions 203381 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r203381 | twilson | 2009-06-25 16:15:11 -0500 (Thu, 25 Jun 2009)
| 11 lines Merged revisions 203380 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r203380 | twilson | 2009-06-25 16:13:10 -0500 (Thu, 25 Jun 2009)
| 4 lines I didn't see that Mark already fixed the underlying
issue! Yay for removing useless code. ........ ................
2009-06-25 21:08 +0000 [r203379] Russell Bryant <russell@digium.com>
* /, main/features.c: Merged revisions 203376 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r203376 | russell | 2009-06-25 16:04:55 -0500 (Thu, 25 Jun 2009)
| 16 lines Merged revisions 203375 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r203375 | russell | 2009-06-25 16:02:18 -0500 (Thu, 25 Jun 2009)
| 9 lines Fix a case where CDR answer time could be before the
start time involving parking. (closes issue #13794) Reported by:
davidw Patches: 13794.patch uploaded by murf (license 17)
13794.patch.160 uploaded by murf (license 17) Tested by: murf,
dbrooks ........ ................
2009-06-25 19:27 +0000 [r203276] Jason Parker <jparker@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 203258 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r203258 | qwell | 2009-06-25 14:22:46 -0500 (Thu, 25 Jun 2009) |
10 lines Unmute when we get a dtmfup (we muted on dtmfdown)
event. This would occasionally cause one-way audio when using
hardware DTMF detection. (closes issue #14761) Reported by:
tzafrir Patches: v1-14761.patch uploaded by dimas (license 88)
Tested by: tzafrir, dimas ........
2009-06-25 16:08 +0000 [r203119] Russell Bryant <russell@digium.com>
* /, channels/chan_sip.c: Merged revisions 203116 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r203116 | russell | 2009-06-25 11:04:10 -0500 (Thu, 25 Jun 2009)
| 18 lines Merged revisions 203115 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r203115 | russell | 2009-06-25 11:02:16 -0500 (Thu, 25 Jun 2009)
| 11 lines Resolve a crash related to a T.38 reinvite race
condition. This change resolves a crash observed locally during
some T.38 testing. A call was set up using a call file, and when
the T.38 reinvite came in, the channel state was still
AST_STATE_DOWN. The reason is explained by a comment in the code
that previously lived in the handling of AST_STATE_RINGING. This
change modifies the logic to handle the same race condition for
any channel state that is not UP. (closes ABE-1895) ........
................
2009-06-24 21:27 +0000 [r203077] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 203037 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r203037 | rmudgett | 2009-06-24 16:08:55 -0500
(Wed, 24 Jun 2009) | 15 lines Merged revisions 203036 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r203036 | rmudgett | 2009-06-24 16:01:43 -0500 (Wed, 24 Jun 2009)
| 8 lines Improved chan_dahdi.conf pritimer error checking. Valid
format is: pritimer=timer_name,timer_value * Fixed segfault if
the ',' is missing. * Completely check the range returned by
pri_timer2idx() to prevent possible access outside array bounds.
........ ................
2009-06-24 18:30 +0000 [r202970] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 202967 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r202967 | mmichelson | 2009-06-24 13:29:10 -0500 (Wed, 24 Jun
2009) | 9 lines Merged revisions 202966 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r202966 | mmichelson | 2009-06-24 13:28:47 -0500 (Wed, 24 Jun
2009) | 3 lines Use the handy UNLINK macro instead of hand-coding
the same thing in-line. ........ ................
2009-06-24 18:11 +0000 [r202928] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 202925 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r202925 |
file | 2009-06-24 15:08:17 -0300 (Wed, 24 Jun 2009) | 2 lines
Ensure the default settings are applied for T.38 when we set it
up for a peer. ........
2009-06-23 23:58 +0000 [r202842] Sean Bright <sean@malleable.com>
* doc/tex/cdrdriver.tex, /, doc/tex/billing.tex: Merged revisions
202840-202841 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r202840 |
seanbright | 2009-06-23 19:53:45 -0400 (Tue, 23 Jun 2009) | 1
line Remove some trailing whitespace before making content
changes. ........ r202841 | seanbright | 2009-06-23 19:57:07
-0400 (Tue, 23 Jun 2009) | 1 line Change some section names in
the CDR tex documentation. ........
2009-06-23 22:47 +0000 [r202805] Russell Bryant <russell@digium.com>
* doc/tex/cdrdriver.tex, /: Merged revisions 202804 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r202804 | russell | 2009-06-23 17:47:26 -0500 (Tue, 23 Jun 2009)
| 2 lines Clean up section hierarchy for the CDR chapter.
........
2009-06-23 22:12 +0000 [r202765] Matthew Fredrickson <creslin@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 202761 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r202761 | mattf | 2009-06-23 17:08:43 -0500 (Tue, 23 Jun 2009) |
1 line I could have sworn I committed this patch ages ago, but...
bug fix with setting NAI properly on linksets in certain
situations. ........
2009-06-23 16:33 +0000 [r202673] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 202672 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r202672 | dvossel | 2009-06-23 11:31:30 -0500 (Tue, 23 Jun 2009)
| 18 lines Merged revisions 202671 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r202671 | dvossel | 2009-06-23 11:28:46 -0500 (Tue, 23 Jun 2009)
| 12 lines MWI NOTIFY contains a wrong URI if Asterisk listens to
non-standard port and transport (closes issue #14659) Reported
by: klaus3000 Patches: patch_chan_sip_fixMWIuri_1.4.txt uploaded
by klaus3000 (license 65) mwi_port-transport_trunk.diff uploaded
by dvossel (license 671) Tested by: dvossel, klaus3000 Review:
https://reviewboard.asterisk.org/r/288/ ........ ................
2009-06-22 20:19 +0000 [r202495-202511] Russell Bryant <russell@digium.com>
* main/channel.c, /: Merged revisions 202497 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r202497 | russell | 2009-06-22 15:11:04 -0500 (Mon, 22 Jun 2009)
| 11 lines Merged revisions 202496 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r202496 | russell | 2009-06-22 15:08:53 -0500 (Mon, 22 Jun 2009)
| 4 lines Report CallerID change during a masquerade. Reported
by: markster ........ ................
* /, channels/chan_sip.c: Merged revisions 202415 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r202415 | russell | 2009-06-22 11:05:08 -0500 (Mon, 22 Jun 2009)
| 9 lines Merged revisions 202414 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r202414 | russell | 2009-06-22 11:00:00 -0500 (Mon, 22 Jun 2009)
| 2 lines Make Polycom subscription type override check more
explicit. ........ ................
2009-06-22 16:31 +0000 [r202473] Sean Bright <sean@malleable.com>
* cdr/cdr_sqlite3_custom.c, /: Merged revisions 202417 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r202417 | seanbright | 2009-06-22 12:09:50 -0400 (Mon, 22 Jun
2009) | 4 lines Fix lock usage in cdr_sqlite3_custom to avoid
potential crashes during reload. Pointed out by Russell while
working on the CEL branch. ........
2009-06-22 15:37 +0000 [r202411] David Vossel <dvossel@digium.com>
* main/loader.c, /, include/asterisk/module.h: Merged revisions
202410 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r202410 |
dvossel | 2009-06-22 10:33:35 -0500 (Mon, 22 Jun 2009) | 5 lines
attempting to load running modules Modules placed in the priority
heap for loading were not properly removed from the linked list.
This resulted in some modules attempting to load twice. ........
2009-06-22 15:17 +0000 [r202340-202346] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 202343 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r202343 | mmichelson | 2009-06-22 09:58:24 -0500 (Mon, 22 Jun
2009) | 36 lines Merged revisions 202341-202342 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r202341 | mmichelson | 2009-06-22 09:42:55 -0500 (Mon, 22 Jun
2009) | 26 lines Fix a situation in which Asterisk would not stop
retransmitting 487s. If a CANCEL were received by Asterisk, we
would send a 487 in response to the original INVITE and a 200 OK
for the CANCEL. If there were a network hiccup which caused the
200 OK and the 487 to be lost, then the UA communicating with
Asterisk may try to retransmit its CANCEL. Asterisk's response to
this used to be to try sending another 487 to the canceled INVITE
and another 200 OK to the CANCEL. The problem here is that the
originally-sent 487 was sent "reliably" meaning that it will be
retransmitted until it is received properly. So when we receive
the second CANCEL it is likely that the first batch of 487s we
sent is still going strong and reaches the UA. The result was
that the second set of 487s would be retransmitted constantly
until the maximum number of retries had been reached. The fix for
this is that if we receive a second CANCEL for an INVITE, then we
cancel the retransmission of the first set of 487s and start a
second set. This causes the dialog to be terminated reasonably.
(closes issue #14584) Reported by: klaus3000 Patches:
14584_v2.patch uploaded by mmichelson (license 60) Tested by:
klaus3000 ........ r202342 | mmichelson | 2009-06-22 09:44:58
-0500 (Mon, 22 Jun 2009) | 3 lines Remove an extra debug line
left from previous commit. ........ ................
* /, channels/chan_sip.c: Merged revisions 202337 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r202337 | mmichelson | 2009-06-22 09:35:09 -0500 (Mon, 22 Jun
2009) | 31 lines Merged revisions 202336 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r202336 | mmichelson | 2009-06-22 09:34:05 -0500 (Mon, 22 Jun
2009) | 25 lines Fix a possible infinite loop in SDP parsing
during glare situation. There was a while loop in
get_ip_and_port_from_sdp which was controlled by a call to
get_sdp_iterate. The loop would exit either if what we were
searching for was found or if the return was NULL. The problem is
that get_sdp_iterate never returns NULL. This means that if what
we were searching for was not present, the loop would run
infinitely. This modification of the loop fixes the problem.
(closes issue #15213) Reported by: schmidts (closes issue #15349)
Reported by: samy (closes issue #14464) Reported by: pj (closes
issue #15345) Reported by: aragon Patches: sip_inf_loop.patch
uploaded by mmichelson (license 60) Tested by: aragon ........
................
2009-06-21 16:16 +0000 [r202261-202265] Russell Bryant <russell@digium.com>
* cdr/cdr_manager.c, /: Merged revisions 202262 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r202262 |
russell | 2009-06-21 11:11:48 -0500 (Sun, 21 Jun 2009) | 2 lines
Fix possibility of crashiness during reload in custom fields
handling. ........
* cdr/cdr_manager.c, /: Merged revisions 202258 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r202258 |
russell | 2009-06-21 11:00:23 -0500 (Sun, 21 Jun 2009) | 2 lines
Standardize return values of load_config() so reload() doesn't
report an error on success. ........
2009-06-20 19:14 +0000 [r202186] Sean Bright <sean@malleable.com>
* /, apps/app_fax.c: Merged revisions 202183 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r202183 |
seanbright | 2009-06-20 15:09:47 -0400 (Sat, 20 Jun 2009) | 5
lines Fix version detection for API changes in spandsp. (closes
issue #15355) Reported by: deuffy ........
2009-06-19 21:08 +0000 [r202007] Matthew Nicholson <mnicholson@digium.com>
* channels/chan_sip.c: Added deadlock protection to
try_suggested_sip_codec in chan_sip.c. Review:
https://reviewboard.asterisk.org/r/287/
2009-06-19 20:26 +0000 [r201995] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c, /: Merged revisions 201994 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r201994 | dvossel | 2009-06-19 15:24:37 -0500
(Fri, 19 Jun 2009) | 14 lines Merged revisions 201993 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r201993 | dvossel | 2009-06-19 15:22:02 -0500 (Fri, 19 Jun 2009)
| 8 lines timestamp was being converted to host order as a short
rather than a long (closes issue #15361) Reported by: ffloimair
Patches: ts_issue.diff uploaded by dvossel (license 671) ........
................
2009-06-19 15:49 +0000 [r201785-201906] Tilghman Lesher <tlesher@digium.com>
* res/res_config_odbc.c, /: Merged revisions 201904 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r201904 | tilghman | 2009-06-19 10:47:55 -0500 (Fri, 19 Jun 2009)
| 4 lines Fix 2 typos and add support for wide character types.
Reported by Benny Amorsen via the asterisk-users mailing list.
http://lists.digium.com/pipermail/asterisk-users/2009-June/233622.html
........
* /, main/features.c: Merged revisions 201829 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r201829 | tilghman | 2009-06-18 19:43:41 -0500 (Thu, 18 Jun 2009)
| 13 lines Merged revisions 201828 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r201828 | tilghman | 2009-06-18 19:40:41 -0500 (Thu, 18 Jun 2009)
| 6 lines If the "h" extension fails, give it another chance in
main/pbx.c. If the "h" extension fails, give it another chance in
main/pbx.c, when it returns from the bridge code. Fixes an issue
where the "h" extension may occasionally not fire, when a Dial is
executed from a Macro. Debugged in #asterisk with user tompaw.
........ ................
* /, apps/Makefile: Merged revisions 201783 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r201783 |
tilghman | 2009-06-18 15:52:36 -0500 (Thu, 18 Jun 2009) | 6 lines
One of the changes in 1.6.1 was to allow app_directory to use
functionality within app_voicemail for directory functions. It is
therefore no longer necessary for app_directory to be linked
against the ODBC libraries (and it never was necessary for
app_directory to be linked against IMAP, though it was). ........
2009-06-18 16:44 +0000 [r201679] David Vossel <dvossel@digium.com>
* channels/misdn/isdn_lib.c, utils/conf2ael.c, main/ast_expr2.c,
utils/stereorize.c, main/ast_expr2f.c, res/ael/ael_lex.c,
utils/ael_main.c, utils/extconf.c, channels/xpmr/xpmr.c,
pbx/pbx_config.c, res/res_config_ldap.c, apps/app_rpt.c,
main/asterisk.c, codecs/gsm/src/gsm_destroy.c, /,
channels/h323/ast_h323.cxx: Merged revisions 201678 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r201678 | dvossel | 2009-06-18 11:37:42 -0500 (Thu, 18 Jun 2009)
| 11 lines fixes some memory leaks and redundant conditions
(closes issue #15269) Reported by: contactmayankjain Patches:
patch.txt uploaded by contactmayankjain (license 740)
memory_leak_stuff.trunk.diff uploaded by dvossel (license 671)
Tested by: contactmayankjain, dvossel ........
2009-06-18 15:40 +0000 [r201614] Russell Bryant <russell@digium.com>
* res/res_musiconhold.c, /: Merged revisions 201610 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r201610 | russell | 2009-06-18 10:27:10 -0500
(Thu, 18 Jun 2009) | 36 lines Merged revisions 201600 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r201600 | russell | 2009-06-18 10:24:31 -0500 (Thu, 18 Jun 2009)
| 29 lines Fix memory corruption and leakage related reloads of
non files mode MoH classes. For Music on Hold classes that are
not files mode, meaning that we are executing an application that
will feed us audio data, we use a thread to monitor the external
application and read audio from it. This thread also makes use of
the MoH class object. In the MoH class destructor, we used
pthread_cancel() to ask the thread to exit. Unfortunately, the
code did not wait to ensure that the thread actually went away.
What needed to be done is a pthread_join() to ensure that the
thread fully cleans up before we proceed. By adding this one
line, we resolve two significant problems: 1) Since the thread
was never joined, it never fully goes away. So, on every reload
of non-files mode MoH, an unused thread was sticking around. 2)
There was a race condition here where the application monitoring
thread could still try to access the MoH class, even though the
thread executing the MoH reload has already destroyed it. (issue
#15109) Reported by: jvandal (issue #15123) Reported by:
axisinternet (issue #15195) Reported by: amorsen (issue AST-208)
........ ................
2009-06-18 15:23 +0000 [r201595] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 201570 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r201570 |
dvossel | 2009-06-18 10:16:05 -0500 (Thu, 18 Jun 2009) | 11 lines
parsing extension correctly from sip register lines If a
transport type was specified, but no extension, parsing of the
extension would return whatever was after the transport rather
than defaulting to 's'. (closes issue #15111) Reported by: ffs
Patches: chan_sip.c_register-parser.patch uploaded by ffs
(license 730) Tested by: ffs, dvossel ........
2009-06-17 21:33 +0000 [r201533] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c, /: Merged revisions 201531 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r201531 | tilghman | 2009-06-17 16:31:39 -0500 (Wed, 17 Jun 2009)
| 7 lines Initialize additional variables, to prevent a possible
crash. (closes issue #15186) Reported by: ajohnson Patches:
20090528__issue15186.diff.txt uploaded by tilghman (license 14)
Tested by: ajohnson ........
2009-06-17 20:12 +0000 [r201461-201465] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 201462 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r201462 |
mmichelson | 2009-06-17 15:10:01 -0500 (Wed, 17 Jun 2009) | 12
lines Fix problem with no audio due to ignoring the SDP. A recent
change to our SDP version comparison made audio not function on
some calls. This was because of a test wherein we were trying to
see if an unsigned value was less than 0. This is a dumb
comparison and arguably the compiler should have warned about it.
Alas, though, it slipped past. Now it's fixed by changing the
variable to be a signed type. Found by several developers. Tested
by mnicholson and dbrooks. ........
* main/channel.c, /: Merged revisions 201458 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r201458 | mmichelson | 2009-06-17 15:04:12 -0500 (Wed, 17 Jun
2009) | 15 lines Merged revisions 201450 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r201450 | mmichelson | 2009-06-17 14:59:31 -0500 (Wed, 17 Jun
2009) | 9 lines Change the datastore traversal in
ast_do_masquerade to use a safe list traversal. It is possible
for datastore fixup functions to remove the datastore from the
list and free it. In particular, the queue_transfer_fixup in
app_queue does this. While I don't yet know of this causing any
crashes, it certainly could. Found while discussing a separate
issue with Brian Degenhardt. ........ ................
2009-06-17 20:01 +0000 [r201447-201454] David Vossel <dvossel@digium.com>
* doc/datastores.txt, /: Merged revisions 201453 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r201453 |
dvossel | 2009-06-17 15:00:51 -0500 (Wed, 17 Jun 2009) | 3 lines
ast_channel_datastore_alloc is no longer used. updating
datastores.txt to reflect that. ........
* apps/app_mixmonitor.c, /: Merged revisions 201445 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r201445 | dvossel | 2009-06-17 14:45:35 -0500
(Wed, 17 Jun 2009) | 25 lines Merged revisions 201423 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r201423 | dvossel | 2009-06-17 14:28:12 -0500 (Wed, 17 Jun 2009)
| 19 lines StopMixMonitor race condition (not giving up file
immediately) StopMixMonitor only indicates to the MixMonitor
thread to stop writing to the file. It does not guarantee that
the recording's file handle is available to the dialplan
immediately after execution. This results in a race condition. To
resolve this, the filestream pointer is placed in a datastore on
the channel. When StopMixMonitor is called, the datastore is
retrieved from the channel and the filestream is closed
immediately before returning to the dialplan. Documentation
indicating the use of StopMixMonitor to free files has been
updated as well. (closes issue #15259) Reported by: travisghansen
Tested by: dvossel Review:
https://reviewboard.asterisk.org/r/283/ ........ ................
2009-06-17 19:49 +0000 [r201446] David Brooks <dbrooks@digium.com>
* /, channels/chan_sip.c: Merged revisions 201381 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r201381 | dbrooks | 2009-06-17 14:15:07 -0500 (Wed, 17 Jun 2009)
| 16 lines Merged revisions 201380 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r201380 | dbrooks | 2009-06-17 13:45:50 -0500 (Wed, 17 Jun 2009)
| 9 lines Checks for NULL sip_pvt pointer in
chan_sip.c->acf_channel_read() Zombie channels could be passed,
and chan_sip.c wasn't checking for it. Could crash Asterisk. Now
checking for NULL pointer. (closes issue #15330) Reported by:
okrief Tested by: dbrooks ........ ................
2009-06-17 15:25 +0000 [r201360] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 201344 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r201344 |
dvossel | 2009-06-17 10:20:26 -0500 (Wed, 17 Jun 2009) | 16 lines
SIP registry ref count error During a sip reload, the list of
sip_registry objects are supposed to be traversed, unlinked, and
destroyed, but destruction never takes place due to a ref
counting error. This causes a memory leak when registry items are
removed from sip.conf and reloaded. While the registries are
removed from the global list, they are not removed from the
scheduler. Because of this, SIP register attempts continue to be
sent out for the item even though it may no longer be in the
.conf. (closes issue #15295) Reported by: amorsen Review:
https://reviewboard.asterisk.org/r/282/ ........
2009-06-17 12:06 +0000 [r201265] Kevin P. Fleming <kpfleming@digium.com>
* /, include/asterisk/linkedlists.h: Merged revisions 201262 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r201262 | kpfleming | 2009-06-17 07:04:17 -0500
(Wed, 17 Jun 2009) | 15 lines Merged revisions 201261 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r201261 | kpfleming | 2009-06-17 07:03:25 -0500 (Wed, 17 Jun
2009) | 9 lines Correct AST_LIST_APPEND_LIST behavior when list
to be appended is empty. When the list to be appended is empty,
and the list to be appended to is *not*, AST_LIST_APPEND_LIST
would actually cause the target list to become broken, and no
longer have a pointer to its last entry. This patch fixes the
problem. (reported by Stanislaw Pitucha on the asterisk-dev
mailing list) ........ ................
2009-06-16 22:30 +0000 [r201224] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 201223 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r201223 |
dvossel | 2009-06-16 17:29:30 -0500 (Tue, 16 Jun 2009) | 2 lines
fix issue with build_contact introduced by the "SIP trasnport
type issues" commit ........
2009-06-16 19:47 +0000 [r200990-201097] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk/frame.h, apps/app_chanspy.c,
apps/app_mixmonitor.c, main/channel.c, main/autoservice.c,
main/frame.c, /, apps/app_meetme.c, main/slinfactory.c,
include/asterisk/linkedlists.h, main/file.c,
include/asterisk/channel.h: Merged revisions 201056 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r201056 | kpfleming | 2009-06-16 13:54:30 -0500
(Tue, 16 Jun 2009) | 18 lines Merged revisions 200991 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r200991 | kpfleming | 2009-06-16 12:05:38 -0500 (Tue, 16 Jun
2009) | 11 lines Improve support for media paths that can
generate multiple frames at once. There are various media paths
in Asterisk (codec translators and UDPTL, primarily) that can
generate more than one frame to be generated when the application
calling them expects only a single frame. This patch addresses a
number of those cases, at least the primary ones to solve the
known problems. In addition it removes the broken TRACE_FRAMES
support, fixes a number of bugs in various frame-related API
functions, and cleans up various code paths affected by these
changes. https://reviewboard.asterisk.org/r/175/ ........
................
* /, configure, autoconf/ast_gcc_attribute.m4, configure.ac: Merged
revisions 201090 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r201090 |
kpfleming | 2009-06-16 14:27:12 -0500 (Tue, 16 Jun 2009) | 5
lines Another minor fix to compiler attribute checking.
Defaulting to 'static' for the function scope was bad... so
remove it. ........
* /, configure, autoconf/ast_gcc_attribute.m4, configure.ac: Merged
revisions 200985 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r200985 |
kpfleming | 2009-06-16 11:32:36 -0500 (Tue, 16 Jun 2009) | 7
lines Fix problems with new compiler attribute checking in
configure script. The last changes to ast_gcc_attribute.m4 caused
some problems checking for various attributes, because the scope
of the symbol the attribute is applied to can be important; this
patch allows the scope to be specified for the check. ........
2009-06-16 16:28 +0000 [r200984] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 200946 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r200946 |
dvossel | 2009-06-16 11:03:30 -0500 (Tue, 16 Jun 2009) | 32 lines
SIP transport type issues What this patch addresses: 1.
ast_sip_ouraddrfor() by default binds to the UDP address/port
reguardless if the sip->pvt is of type UDP or not. Now when no
remapping is required, ast_sip_ouraddrfor() checks the sip_pvt's
transport type, attempting to set the address and port to the
correct TCP/TLS bindings if necessary. 2. It is not necessary to
send the port number in the Contact header unless the port is
non-standard for the transport type. This patch fixes this and
removes the todo note. 3. In sip_alloc(), the default dialog
built always uses transport type UDP. Now sip_alloc() looks at
the sip_request (if present) and determines what transport type
to use by default. 4. When changing the transport type of a
sip_socket, the file descriptor must be set to -1 and in some
cases the tcptls_session's ref count must be decremented and set
to NULL. I've encountered several issues associated with this
process and have created a function, set_socket_transport(), to
handle the setting of the socket type. (closes issue #13865)
Reported by: st Patches: dont_add_port_if_tls.patch uploaded by
Kristijan (license 753) 13865.patch uploaded by mmichelson
(license 60) tls_port_v5.patch uploaded by vrban (license 756)
transport_issues.diff uploaded by dvossel (license 671) Tested
by: mmichelson, Kristijan, vrban, jmacz, dvossel Review:
https://reviewboard.asterisk.org/r/278/ ........
2009-06-16 16:05 +0000 [r200948] Michiel van Baak <michiel@vanbaak.info>
* apps/app_voicemail.c, /: Merged revisions 200943 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r200943 | mvanbaak | 2009-06-16 17:51:36 +0200 (Tue, 16 Jun 2009)
| 9 lines add FILE_STORAGE to Voicemail Build Options Voicemail
can only use one storage module at the moment. Because it's
unclear that selecting one of the storage modules in menuselect
will disable filesystem storage we now have a FILE_STORAGE option
that conflicts with the other modules. (closes issue #15333)
........
2009-06-16 12:55 +0000 [r200842] Eliel C. Sardanons <eliels@gmail.com>
* res/res_smdi.c, /: Merged revisions 200841 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r200841 |
eliel | 2009-06-16 08:32:00 -0400 (Tue, 16 Jun 2009) | 6 lines
Show the interface name on error, if it is not found. If the
smdiport specified is not found, show the interface name instead
of '(null)'. ........
2009-06-16 02:41 +0000 [r200807] Moises Silva <moises.silva@gmail.com>
* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged
revisions 200799 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r200799 |
moy | 2009-06-15 21:24:30 -0500 (Mon, 15 Jun 2009) | 2 lines keep
backwards compatible chan_dahdi with older openr2 versions by not
using the new skip category feature unless supported ........
2009-06-16 01:30 +0000 [r200690-200765] Kevin P. Fleming <kpfleming@digium.com>
* /, configure, include/asterisk/autoconfig.h.in,
autoconf/ast_gcc_attribute.m4: Merged revisions 200764 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r200764 | kpfleming | 2009-06-15 20:28:08 -0500 (Mon, 15
Jun 2009) | 11 lines Ensure that configure-script testing for
compiler attributes actually works. The configure script tests
for compiler attributes didn't actually enable enough warnings or
provide a proper test harness to determine whether the compiler
supports the attribute in question or not; this caused gcc 4.1 to
report that it supports 'weakref', but it doesn't actually
support it in the way that is needed for our optional API
mechanism. The new configure script test will properly
distinguish between full support and partial support for this
attribute, among others. ........
* CHANGES, /: Merged revisions 200726 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r200726 |
kpfleming | 2009-06-15 20:03:22 -0500 (Mon, 15 Jun 2009) | 6
lines Document the new automatic 'ignoresdpversion' behavior.
Asterisk will now automatically ignore incorrect incoming SDP
version numbers when necessary to complete a T.38 re-INVITE
operation. ........
* /, channels/chan_sip.c: Merged revisions 200689 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r200689 |
kpfleming | 2009-06-15 15:42:38 -0500 (Mon, 15 Jun 2009) | 11
lines Accept T.38 re-INVITE responses with invalid SDP versions.
This commit changes the 'incoming SDP version' check logic a bit
more; when 'ignoresdpversion' is *not* set for a peer, if we
initiate a re-INVITE to switch to T.38, we'll always accept the
peer's SDP response, even if they don't properly increment the
SDP version number as they should. If this situation occurs, a
warning message will be generated suggesting that the peer's
configuration be changed to include the 'ignoresdpversion'
configuration option (although ideally they'd fix their SIP
implementation to be RFC compliant). AST-221 ........
2009-06-15 15:23 +0000 [r200517] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 200514 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r200514 | mmichelson | 2009-06-15 10:22:11 -0500 (Mon, 15 Jun
2009) | 11 lines Merged revisions 200513 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r200513 | mmichelson | 2009-06-15 10:21:46 -0500 (Mon, 15 Jun
2009) | 5 lines Add INFO to our allowed methods so that endpoints
know they may send it to us. AST-223 ........ ................
2009-06-14 06:33 +0000 [r200512] Moises Silva <moises.silva@gmail.com>
* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /,
build_tools/menuselect-deps.in: Merged revisions 200477 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r200477 | moy | 2009-06-14 01:13:48 -0500 (Sun, 14 Jun
2009) | 3 lines added openr2 to menuselect-deps.in, recent commit
in menuselect made me realize this was never done but was working
anyways also added support for skip category request feature of
openr2 and updated chan_dahdi.conf.sample ........
2009-06-12 19:08 +0000 [r200364] Mark Michelson <mmichelson@digium.com>
* main/channel.c, /: Merged revisions 200361 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r200361 | mmichelson | 2009-06-12 14:07:51 -0500 (Fri, 12 Jun
2009) | 16 lines Merged revisions 200360 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r200360 | mmichelson | 2009-06-12 14:06:41 -0500 (Fri, 12 Jun
2009) | 10 lines Suppress a warning message and give a better
return code when generating inband ringing after a call is
answered. (closes issue #15158) Reported by: madkins Patches:
15158.patch uploaded by mmichelson (license 60) Tested by:
madkins ........ ................
2009-06-12 02:20 +0000 [r200198-200255] Sean Bright <sean@malleable.com>
* contrib/init.d/rc.debian.asterisk, /: Merged revisions 200254 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r200254 | seanbright | 2009-06-11 22:20:19 -0400 (Thu,
11 Jun 2009) | 5 lines Call chgrp instead of chown when setting
run directory group ownership. (issue #13153) Reported by:
pabelanger ........
* Makefile, /: Merged revisions 199781 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r199781 |
seanbright | 2009-06-09 14:08:53 -0400 (Tue, 09 Jun 2009) | 2
lines Fix all of the parallel build warnings issued when running
make -j#. ........
* /: Undo block of revision 199782 (will be merging it momentarily)
2009-06-11 21:35 +0000 [r200172] Terry Wilson <twilson@digium.com>
* main/rtp.c: Don't access rtp->rtcp->* if rtp->rtcp is null
2009-06-11 21:18 +0000 [r200154] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 200146 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r200146 |
mmichelson | 2009-06-11 16:17:14 -0500 (Thu, 11 Jun 2009) | 5
lines Fix a crash due to a potentially NULL p->options. Thanks to
mnicholson for pointing it out. ........
2009-06-11 Leif Madsen <lmadsen@digium.com>
* Release Asterisk 1.6.2.0-beta3
2009-06-11 12:19 +0000 [r200051] Leif Madsen <lmadsen@digium.com>
* build_tools/make_version_h, /, build_tools/make_version_c: Merged
revisions 200039 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r200039 |
lmadsen | 2009-06-11 08:15:09 -0400 (Thu, 11 Jun 2009) | 8 lines
Fix path for .flavor and .version (issue #14737) Reported by:
davidw Patches: flavor.patch uploaded by davidw (license 780)
Tested by: davidw ........
2009-06-10 20:37 +0000 [r199998] David Brooks <dbrooks@digium.com>
* main/pbx.c, /: Fixes the argument order in definition of
new_find_extension(). In the definition of new_find_extension(),
the arguments 'callerid' and 'label' were swapped. The prototype
declaration and all calls to the function are ordered 'callerid'
then 'label', but the function itself was ordered 'label' then
'callerid'. (closes issue #15303) Reported by: JimDickenson
2009-06-10 20:18 +0000 [r199966] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 199958 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r199958 |
mmichelson | 2009-06-10 15:15:48 -0500 (Wed, 10 Jun 2009) | 6
lines Only try to use the invite_branch on outgoing INVITEs with
auth credentials. I have added a comment to the code to help ease
understanding of the logic here as well. ........
2009-06-10 16:13 +0000 [r199860] Sean Bright <sean.bright@gmail.com>
* include/asterisk/utils.h, /: Merged revisions 199857 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r199857 | seanbright | 2009-06-10 12:10:23 -0400
(Wed, 10 Jun 2009) | 9 lines Merged revisions 199856 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r199856 | seanbright | 2009-06-10 12:08:35 -0400 (Wed,
10 Jun 2009) | 2 lines __WORDSIZE is not available on all
platforms, so use sizeof(void *) instead. ........
................
2009-06-09 20:48 +0000 [r199744-199819] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 199818 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r199818 |
dvossel | 2009-06-09 15:47:57 -0500 (Tue, 09 Jun 2009) | 11 lines
CLI NOTIFY sending wrong transport type. SIP's cli NOTIFY command
only used UDP rather than copying the transport type from the
peer. (closes issue #15283) Reported by: jthurman Patches:
sip-notify-tcp-svn199728.patch uploaded by jthurman (license 614)
Tested by: jthurman, dvossel ........
* main/loader.c, /, res/res_timing_pthread.c,
include/asterisk/module.h, res/res_timing_dahdi.c,
res/res_timing_timerfd.c: Merged revisions 199743 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r199743 | dvossel | 2009-06-09 11:22:04 -0500 (Tue, 09 Jun 2009)
| 11 lines module load priority This patch adds the option to
give a module a load priority. The value represents the order in
which a module's load() function is initialized. The lower the
value, the higher the priority. The value is only checked if the
AST_MODFLAG_LOAD_ORDER flag is set. If the AST_MODFLAG_LOAD_ORDER
flag is not set, the value will never be read and the module will
be given the lowest possible priority on load. Since some modules
are reliant on a timing interface, the timing modules have been
given a high load priorty. (closes issue #15191) Reported by:
alecdavis Tested by: dvossel Review:
https://reviewboard.asterisk.org/r/262/ ........
2009-06-08 19:39 +0000 [r199634] Sean Bright <sean.bright@gmail.com>
* include/asterisk/utils.h, /: Merged revisions 199630 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r199630 | seanbright | 2009-06-08 15:33:09 -0400
(Mon, 08 Jun 2009) | 32 lines Merged revisions 199626,199628 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r199626 | seanbright | 2009-06-08 15:24:32 -0400 (Mon, 08 Jun
2009) | 21 lines Increase the size of our thread stack on 64 bit
processors. We were setting the stack size for each thread to
240KB regardless of architecture, which meant that in some
scenarios we actually had less available stack space on 64 bit
processors (pointers use 8 bytes instead of 4). So now we
calculate the stack size we reserve based on the platform's
__WORDSIZE, which gives us: 32 bit -> 240KB 64 bit -> 496KB 128
bit -> 1008KB (that's right, we're ready for 128 bit processors)
Patch typed by me but written by several members of
#asterisk-dev, including Kevin, Tilghman, and Qwell. (closes
issue #14932) Reported by: jpiszcz Patches:
06052009_issue14932.patch uploaded by seanbright (license 71)
Tested by: seanbright ........ r199628 | seanbright | 2009-06-08
15:28:33 -0400 (Mon, 08 Jun 2009) | 2 lines Fix a typo in the
stack size calculation just introduced. ........ ................
2009-06-08 17:42 +0000 [r199591] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Recorded merge of revisions 199588 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r199588 | mmichelson | 2009-06-08 12:32:04 -0500 (Mon,
08 Jun 2009) | 9 lines Fix a deadlock that could occur when
setting rtp stats on SIP calls. (closes issue #15143) Reported
by: cristiandimache Patches: 15143.patch uploaded by mmichelson
(license 60) Tested by: cristiandimache ........
2009-06-06 21:39 +0000 [r199369] Russell Bryant <russell@digium.com>
* Makefile, /: Merged revisions 199368 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r199368 |
russell | 2009-06-06 16:38:54 -0500 (Sat, 06 Jun 2009) | 2 lines
Switch from "echo -n" to printf. On my mac, the -n was just
getting printed out. ........
2009-06-05 21:25 +0000 [r199299] David Vossel <dvossel@digium.com>
* include/asterisk/devicestate.h, /, main/devicestate.c: Merged
revisions 199298 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r199298 | dvossel | 2009-06-05 16:21:22 -0500 (Fri, 05 Jun 2009)
| 21 lines Merged revisions 199297 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r199297 | dvossel | 2009-06-05 16:19:56 -0500 (Fri, 05 Jun 2009)
| 14 lines Fixes issue with hints giving unexpected results.
Hints with two or more devices that include ONHOLD gave
unexpected results. (closes issue #15057) Reported by:
p_lindheimer Patches: onhold_trunk.diff uploaded by dvossel
(license 671) pbx.c.1.4.patch uploaded by p (license 558)
devicestate.c.trunk.patch uploaded by p (license 671) Tested by:
p_lindheimer, dvossel Review:
https://reviewboard.asterisk.org/r/254/ ........ ................
2009-06-05 13:52 +0000 [r199230] Mark Michelson <mmichelson@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 199227 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r199227 | mmichelson | 2009-06-05 08:51:08 -0500 (Fri, 05 Jun
2009) | 14 lines Correct "dahdi show channels" output when
specifying a group. Since a DAHDI channel may belong to multiple
groups, we need to use a bitwise and instead of equivalence to
determine whether to display the channel information. (closes
issue #15248) Reported by: gentian Patches: 15248.patch uploaded
by mmichelson (license 60) Tested by: gentian ........
2009-06-04 19:15 +0000 [r199140] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c, /: Merged revisions 199139 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r199139 | dvossel | 2009-06-04 14:10:16 -0500
(Thu, 04 Jun 2009) | 9 lines Merged revisions 199138 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r199138 | dvossel | 2009-06-04 14:00:15 -0500 (Thu, 04
Jun 2009) | 3 lines Additional updates to AST-2009-001 ........
................
2009-06-04 14:53 +0000 [r199054] Sean Bright <sean.bright@gmail.com>
* include/asterisk/_private.h, main/asterisk.c, main/loader.c, /:
Merged revisions 199051 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r199051 | seanbright | 2009-06-04 10:31:24 -0400 (Thu, 04 Jun
2009) | 47 lines Merged revisions 199022 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r199022 | seanbright | 2009-06-04 10:14:57 -0400 (Thu, 04 Jun
2009) | 40 lines Safely handle AMI connections/reload requests
that occur during startup. During asterisk startup, a lock on the
list of modules is obtained by the primary thread while each
module is initialized. Issue 13778 pointed out a problem with
this approach, however. Because the AMI is loaded before other
modules, it is possible for a module reload to be issued by a
connected client (via Action: Command), causing a deadlock. The
resolution for 13778 was to move initialization of the manager to
happen after the other modules had already been lodaded. While
this fixed this particular issue, it caused a problem for users
(like FreePBX) who call AMI scripts via an #exec in a
configuration file (See issue 15189). The solution I have come up
with is to defer any reload requests that come in until after the
server is fully booted. When a call comes in to ast_module_reload
(from wherever) before we are fully booted, the request is added
to a queue of pending requests. Once we are done booting up, we
then execute these deferred requests in turn. Note that I have
tried to make this a bit more intelligent in that it will not
queue up more than 1 request for the same module to be reloaded,
and if a general reload request comes in ('module reload') the
queue is flushed and we only issue a single deferred reload for
the entire system. As for how this will impact existing
installations - Before 13778, a reload issued before module
initialization was completed would result in a deadlock. After
13778, you simply couldn't connect to the manager during startup
(which causes problems with #exec-that-calls-AMI configuration
files). I believe this is a good general purpose solution that
won't negatively impact existing installations. (closes issue
#15189) (closes issue #13778) Reported by: p_lindheimer Patches:
06032009_15189_deferred_reloads.diff uploaded by seanbright
(license 71) Tested by: p_lindheimer, seanbright Review:
https://reviewboard.asterisk.org/r/272/ ........ ................
2009-06-03 15:24 +0000 [r198827-198886] David Vossel <dvossel@digium.com>
* main/channel.c, /, main/features.c, include/asterisk/channel.h:
Merged revisions 198856 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r198856 |
dvossel | 2009-06-02 16:17:49 -0500 (Tue, 02 Jun 2009) | 10 lines
Generic call forward api, ast_call_forward() The function
ast_call_forward() forwards a call to an extension specified in
an ast_channel's call_forward string. After an ast_channel is
called, if the channel's call_forward string is set this function
can be used to forward the call to a new channel and terminate
the original one. I have included this api call in both
channel.c's ast_request_and_dial() and feature.c's
feature_request_and_dial(). App_dial and app_queue already
contain call forward logic specific for their application and
options. (closes issue #13630) Reported by: festr Review:
https://reviewboard.asterisk.org/r/271/ ........
* channels/chan_iax2.c, /: Merged revisions 198824 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r198824 | dvossel | 2009-06-02 12:55:35 -0500 (Tue, 02 Jun 2009)
| 8 lines fixes issue with channels not going down after transfer
Iax2 currently does not support native bridging if the timeoutms
value is set. We check for that in iax2_bridge, but then set
timeoutms to 0 by default. If the timeoutms is not provided it is
set to -1. By setting timeoutms to 0 it is processed causing a
bridging retry loop. (closes issue #15216) Reported by: oxymoron
Tested by: dvossel ........
2009-06-02 13:51 +0000 [r198794] Joshua Colp <jcolp@digium.com>
* configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions
198791 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r198791 |
file | 2009-06-02 10:48:06 -0300 (Tue, 02 Jun 2009) | 5 lines
Correct documentation for the register line, specifically where
the domain should be specified. (closes issue #14367) Reported
by: Nick_Lewis ........
2009-06-01 21:04 +0000 [r198730] Russell Bryant <russell@digium.com>
* channels/iax2-parser.c, /: Merged revisions 198729 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r198729 | russell | 2009-06-01 16:03:18 -0500 (Mon, 01 Jun 2009)
| 2 lines Tell the IAX2 parser about more control frame types.
........
2009-06-01 18:44 +0000 [r198629] Tilghman Lesher <tlesher@digium.com>
* /, contrib/scripts/meetme.sql: Merged revisions 198626 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r198626 | tilghman | 2009-06-01 13:40:35 -0500 (Mon, 01
Jun 2009) | 2 lines Add information for new meetme realtime
fields ........
2009-05-31 17:53 +0000 [r198471] Tilghman Lesher <tlesher@digium.com>
* /, funcs/func_strings.c: Merged revisions 198470 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r198470 | tilghman | 2009-05-31 12:52:28 -0500 (Sun, 31 May 2009)
| 2 lines Fix documentation for FIELDQTY. ........
2009-05-31 01:48 +0000 [r198440] Eliel C. Sardanons <eliels@gmail.com>
* /, res/res_timing_dahdi.c: Merged revisions 198437 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r198437 | eliel | 2009-05-30 21:22:15 -0400 (Sat, 30 May 2009) |
11 lines Avoid a crash when res_timing_dahdi is unloaded but
wasn't properly loaded. if dahdi_test_timer() fails,
timing_funcs_handle remains NULL causing a crash when calling
ast_unregister_timing_interface() with a NULL pointer. (closes
issue #15234) Reported by: eliel Patches: timing_dahdi1.diff
uploaded by eliel (license 64) ........
2009-05-31 01:21 +0000 [r198436] Russell Bryant <russell@digium.com>
* res/res_smdi.c, /: Merged revisions 198312 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r198312 | russell | 2009-05-29 22:43:23 -0500 (Fri, 29 May 2009)
| 12 lines Merged revisions 198311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r198311 | russell | 2009-05-29 22:42:46 -0500 (Fri, 29 May 2009)
| 5 lines Fix a crash that occurred when MWI SMDI messages
expired. (closes issue #14561) Reported by: cmoss28 ........
................
2009-05-30 20:22 +0000 [r198297-198397] Sean Bright <sean.bright@gmail.com>
* res/res_jabber.c, /: Merged revisions 198375 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r198375 |
seanbright | 2009-05-30 16:11:33 -0400 (Sat, 30 May 2009) | 13
lines Properly terminate the receive buffer before sending to
iksemel. aji_io_recv takes the maximum number of bytes to read
(instead of the total buffer size), so we have to subtract 1 from
our buffer size. Without this, when we receive packets that are
larger than our buffer, iksemel will choke and things get wonky.
(closes issue #15232) Reported by: lp0 Patches:
05302009_res_jabber.c.patch uploaded by seanbright (license 71)
Tested by: seanbright, lp0 ........
* res/res_jabber.c, /: Merged revisions 198371 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r198371 | seanbright | 2009-05-30 15:38:58 -0400 (Sat, 30 May
2009) | 19 lines Merged revisions 198370 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r198370 | seanbright | 2009-05-30 15:36:20 -0400 (Sat, 30 May
2009) | 12 lines Properly terminate AMI JabberSend response
messages. The response message (either Error or Success) needs an
extra trailing \r\n after the fields to inform the client that
the message is complete. (closes issue #14876) Reported by: srt
Patches: 05302009_1.4_res_jabber.c.diff uploaded by seanbright
(license 71) asterisk_14876.patch uploaded by srt (license 378)
trunk-14876-2.diff uploaded by phsultan (license 73) ........
................
* apps/app_dial.c, /: Merged revisions 198285 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r198285 | seanbright | 2009-05-29 23:26:06 -0400 (Fri, 29 May
2009) | 15 lines Merged revisions 198251 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r198251 | seanbright | 2009-05-29 22:46:41 -0400 (Fri, 29 May
2009) | 8 lines Treat an empty FORWARD_CONTEXT the same way we
treat a missing one. (closes issue #15056) Reported by:
p_lindheimer Patches: 05292009_bug15056.diff uploaded by
seanbright (license 71) Tested by: p_lindheimer ........
................
2009-05-30 02:35 +0000 [r198250] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 198248 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r198248 |
file | 2009-05-29 23:31:48 -0300 (Fri, 29 May 2009) | 2 lines
When removing all packets from a dialog we also need to free the
data if present. ........
2009-05-29 23:05 +0000 [r198148-198188] Russell Bryant <russell@digium.com>
* /, configs/modules.conf.sample: Merged revisions 198186 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r198186 | russell | 2009-05-29 18:04:31 -0500 (Fri, 29
May 2009) | 2 lines Suggesting that only a single timing module
be loaded is no longer necessary. ........
* /, res/res_timing_pthread.c: Merged revisions 198183 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r198183 | russell | 2009-05-29 17:33:31 -0500 (Fri, 29 May 2009)
| 2 lines Improve handling of trying to ACK too many timer
expirations. ........
* /, res/res_timing_pthread.c: Merged revisions 198146 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r198146 | russell | 2009-05-29 15:06:59 -0500 (Fri, 29 May 2009)
| 38 lines Resolve issues with choppy sound when using
res_timing_pthread. The situation that caused this problem was
when continuous mode was being turned on and off while a rate was
set for a timing interface. A very easy way to replicate this bug
was to do a Playback() from behind a Local channel. In this
scenario, a rate gets set on the channel for doing file playback.
At the same time, continuous mode gets turned on and off about
every 20 ms as frames get queued on to the PBX side channel from
the other side of the Local channel. Essentially, this module
treated continuous mode and a set rate as mutually exclusive
states for the timer to be in. When I dug deep enough, I observed
the following pattern: 1) Set timer to tick every 20 ms. 2) Wait
almost 20 ms ... 3) Continuous mode gets turned on for a queued
up frame 4) Continuous mode gets turned off 5) The timer goes
back to its tick per 20 ms. state but starts counting at 0 ms. 6)
Goto step 2. Sometimes, res_timing_pthread would make it 20 ms
and produce a timer tick, but not most of the time. This is what
produced the choppy sound (or sometimes no sound at all). Now,
the module treats continuous mode and a set rate as completely
independent timer modes. They can be enabled and disabled
independently of each other and things work as expected. (closes
issue #14412) Reported by: dome Patches: issue14412.diff.txt
uploaded by russell (license 2) issue14412-1.6.1.0.diff.txt
uploaded by russell (license 2) Tested by: DennisD, russell
........
2009-05-29 19:26 +0000 [r198111] Eliel C. Sardanons <eliels@gmail.com>
* CREDITS, /: Merged revisions 198083 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r198083 |
eliel | 2009-05-29 15:18:35 -0400 (Fri, 29 May 2009) | 3 lines
Apply anti-spam obfuscation to an email address. ........
2009-05-29 19:14 +0000 [r198075] Matthew Nicholson <mnicholson@digium.com>
* main/cdr.c, main/channel.c, /, include/asterisk/cdr.h: Merged
revisions 198072 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r198072 | mnicholson | 2009-05-29 14:04:24 -0500 (Fri, 29 May
2009) | 21 lines Merged revisions 198068 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r198068 | mnicholson | 2009-05-29 13:53:01 -0500 (Fri, 29 May
2009) | 15 lines Use AST_CDR_NOANSWER instead of AST_CDR_NULL as
the default CDR disposition. This change also involves the
addition of an AST_CDR_FLAG_ORIGINATED flag that is used on
originated channels to distinguish: them from dialed channels.
(closes issue #12946) Reported by: meral Patches: null-cdr2.diff
uploaded by mnicholson (license 96) Tested by: mnicholson,
dbrooks (closes issue #15122) Reported by: sum Tested by: sum
........ ................
2009-05-29 18:40 +0000 [r198066] Joshua Colp <jcolp@digium.com>
* /, main/file.c: Merged revisions 198064 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r198064 |
file | 2009-05-29 15:39:04 -0300 (Fri, 29 May 2009) | 2 lines Fix
a memory leak of the write buffer when writing a file. ........
2009-05-29 18:18 +0000 [r198008] Sean Bright <sean.bright@gmail.com>
* Makefile, /: Merged revisions 198000 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r198000 | seanbright | 2009-05-29 14:15:15 -0400 (Fri, 29 May
2009) | 15 lines Merged revisions 197998 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r197998 | seanbright | 2009-05-29 14:14:12 -0400 (Fri, 29 May
2009) | 8 lines Fix 'make config' target for Slackware. There was
a missing semi-colon after the echo statement in the Makefile
that was causing problems for some users. Fix suggested by
reporter. (closes issue #15225) Reported by: pdavis ........
................
2009-05-29 16:29 +0000 [r197994] Russell Bryant <russell@digium.com>
* /, res/res_timing_pthread.c: Merged revisions 197960 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r197960 | russell | 2009-05-29 11:15:30 -0500 (Fri, 29 May 2009)
| 2 lines Trim trailing whitespace so that I can work on this bug
without it bothering me. :-) ........
2009-05-28 23:54 +0000 [r197894] Leif Madsen <lmadsen@digium.com>
* apps/app_mixmonitor.c, /: Merged revisions 197828 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r197828 | lmadsen | 2009-05-28 18:04:00 -0400 (Thu, 28 May 2009)
| 8 lines Update documentation in MixMonitor. Updated the
MixMonitor documentation for the 'b' option so that it is more
obvious that you must not optimize away the Local channel when
using this option. (closes issue #14829) Reported by: licedey
Tested by: mmichelson, licedey, lmadsen ........
2009-05-28 18:50 +0000 [r197703] Joshua Colp <jcolp@digium.com>
* channels/chan_iax2.c, /: Merged revisions 197697 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r197697 | file | 2009-05-28 15:45:11 -0300 (Thu, 28 May 2009) | 2
lines Fix a bug where the trunkmtu setting was not set to the
default value of 1240 on load but was on reload. ........
2009-05-28 16:15 +0000 [r197625] Eliel C. Sardanons <eliels@gmail.com>
* /, channels/chan_sip.c: Merged revisions 197621 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r197621 | eliel | 2009-05-28 12:01:48 -0400 (Thu, 28 May 2009) |
19 lines Merged revisions 197562 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r197562 | eliel | 2009-05-28 11:21:32 -0400 (Thu, 28 May 2009) |
13 lines Use the address we already know when reloading a peer
with nat=yes. If we already have an address for a peer, and we
are reloading the sip configuration, try to use that address to
contact the peer, instead of getting it from the Contact. (closes
issue #15194) Reported by: ibc Patches: sip.patch uploaded by
eliel (license 64) Tested by: manwe ........ ................
2009-05-28 15:44 +0000 [r197548-197619] Mark Michelson <mmichelson@digium.com>
* main/rtp.c, /, channels/chan_sip.c, include/asterisk/rtp.h:
Merged revisions 197606 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r197606 | mmichelson | 2009-05-28 10:32:19 -0500 (Thu, 28 May
2009) | 22 lines Recorded merge of revisions 197588 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r197588 | mmichelson | 2009-05-28 10:27:49 -0500 (Thu,
28 May 2009) | 16 lines Allow for media to arrive from an
alternate source when responding to a reinvite with 491. When we
receive a SIP reinvite, it is possible that we may not be able to
process the reinvite immediately since we have also sent a
reinvite out ourselves. The problem is that whoever sent us the
reinvite may have also sent a reinvite out to another party, and
that reinvite may have succeeded. As a result, even though we are
not going to accept the reinvite we just received, it is
important for us to not have problems if we suddenly start
receiving RTP from a new source. The fix for this is to grab the
media source information from the SDP of the reinvite that we
receive. This information is passed to the RTP layer so that it
will know about the alternate source for media. Review:
https://reviewboard.asterisk.org/r/252 ........ ................
* main/audiohook.c, apps/app_chanspy.c, /,
include/asterisk/audiohook.h: Merged revisions 197543 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r197543 | mmichelson | 2009-05-28 09:58:06 -0500
(Thu, 28 May 2009) | 27 lines Merged revisions 197537 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r197537 | mmichelson | 2009-05-28 09:49:13 -0500 (Thu, 28 May
2009) | 21 lines Add flags to chanspy audiohook so that audio
stays in sync. There are two flags being added to the chanspy
audiohook here. One is the pre-existing
AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set, we ensure that
the read and write slinfactories on the audiohook do not skew
beyond a certain tolerance. In addition, there is a new audiohook
flag added here, AST_AUDIOHOOK_SMALL_QUEUE. With this flag set,
we do not allow for a slinfactory to build up a substantial
amount of audio before flushing it. For this particular issue,
this means that the person spying on the call will hear the
conversations in real time with very little delay in the audio.
(closes issue #13745) Reported by: geoffs Patches: 13745.patch
uploaded by mmichelson (license 60) Tested by: snblitz ........
................
2009-05-28 14:56 +0000 [r197471-197542] Joshua Colp <jcolp@digium.com>
* /, main/utils.c: Merged revisions 197538 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r197538 |
file | 2009-05-28 11:51:43 -0300 (Thu, 28 May 2009) | 5 lines Fix
a bug in stringfields where it did not actually free the pools of
memory. (closes issue #15074) Reported by: pj ........
* /, channels/chan_sip.c: Merged revisions 197467 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r197467 | file | 2009-05-28 10:47:45 -0300 (Thu, 28 May 2009) |
15 lines Merged revisions 197466 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r197466 | file | 2009-05-28 10:44:58 -0300 (Thu, 28 May 2009) | 8
lines Fix a bug where the flag indicating the presence of rport
would get overwritten by the nat setting. The presence of rport
is now stored as a separate flag. Once the dialog is setup and
authenticated (or it passes through unauthenticated) the proper
nat flag is set. (closes issue #13823) Reported by: dimas
........ ................
2009-05-28 11:40 +0000 [r197441] Gavin Henry <ghenry@suretecsystems.com>
* contrib/scripts/asterisk.ldap-schema,
contrib/scripts/asterisk.ldif, doc/ldap.txt,
configs/res_ldap.conf.sample: issue #15155 and issue #15156 from
trunk
2009-05-27 23:49 +0000 [r197375] Tilghman Lesher <tlesher@digium.com>
* /, main/xml.c: Merged revisions 197374 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r197374 |
tilghman | 2009-05-27 18:48:15 -0500 (Wed, 27 May 2009) | 2 lines
Revert commit 192032. This define is needed on Mac OS X. ........
2009-05-27 22:23 +0000 [r197336] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk/agi.h, /: Merged revisions 197335 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r197335 | kpfleming | 2009-05-27 17:21:53 -0500 (Wed, 27 May
2009) | 3 lines Ensure that this header includes xmldoc.h, since
it depends on it. ........
2009-05-27 20:11 +0000 [r197263] Sean Bright <sean.bright@gmail.com>
* Makefile, /: Merged revisions 197260 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r197260 |
seanbright | 2009-05-27 16:08:16 -0400 (Wed, 27 May 2009) | 6
lines Use bash explicitly when calling build_tools/mkpkgconfig
from the Makefile. Since we use bashisms in
build_tools/mkpkgconfig, we should call on bash explicitly when
running from the Makefile, otherwise we get errors during a 'make
install.' (closes issue #15209) Reported by: seandarcy ........
2009-05-27 19:30 +0000 [r197247] Tilghman Lesher <tlesher@digium.com>
* /, funcs/func_cut.c: Recorded merge of revisions 197209 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r197209 | tilghman | 2009-05-27 14:20:56 -0500
(Wed, 27 May 2009) | 12 lines Recorded merge of revisions 197194
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r197194 | tilghman | 2009-05-27 14:09:42 -0500 (Wed, 27 May 2009)
| 5 lines Use a different determinator on whether to print the
delimiter, since leading fields may be blank. (closes issue
#15208) Reported by: ramonpeek Patch by me, though inspired in
part by a patch from ramonpeek ........ ................
2009-05-27 17:28 +0000 [r197176] Jeff Peeler <jpeeler@digium.com>
* main/channel.c, include/asterisk/channel.h: Fix broken attended
transfers The bridge was terminating immediately after the
attended transfer was completed. The problem was because upon
reentering ast_channel_bridge nexteventts was checked to see if
it was set and if so could possibly return AST_BRIDGE_COMPLETE.
(closes issue #15183) Reported by: andrebarbosa Tested by:
andrebarbosa, tootai, loloski
2009-05-27 16:12 +0000 [r196950-197092] Sean Bright <sean.bright@gmail.com>
* configs/smdi.conf.sample, configs/extensions.conf.sample,
configs/sla.conf.sample, configs/chan_dahdi.conf.sample, /,
configs/vpb.conf.sample: Merged revisions 197089 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r197089 | seanbright | 2009-05-27 12:07:57 -0400 (Wed, 27 May
2009) | 6 lines Fix references to /etc/dahdi/system.conf and
/etc/asterisk/chan_dahdi.conf in the sample configuration files.
(closes issue #15207) Reported by: seandarcy ........
* /, channels/chan_alsa.c: Merged revisions 196988 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r196988 | seanbright | 2009-05-27 09:02:54 -0400 (Wed, 27 May
2009) | 9 lines Display an error message when chan_alsa fails to
load due to a missing or inaccessible configuration file. Before
this change, when chan_alsa failed to load due to a missing or
inaccessible configuration file, no message would be displayed.
With this change, when chan_alsa fails to load due to a missing
or inaccessible configuration file, a message will be displayed.
(closes issue #14760) Reported by: Nick_Lewis Patches:
chan_alsa.c-confload.patch uploaded by Nick (license 657)
........
* main/xmldoc.c, /: Merged revisions 196948 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r196948 |
seanbright | 2009-05-26 18:43:21 -0400 (Tue, 26 May 2009) | 8
lines Reset the terminal to the correct fg/bg after XML
documenation is rendered. (closes issue #15200) Reported by:
ajohnson Patches: 05262009_xmldoc.patch uploaded by seanbright
(license 71) Tested by: ajohnson ........
* main/manager.c, /: Merged revisions 196945 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r196945 |
seanbright | 2009-05-26 18:38:05 -0400 (Tue, 26 May 2009) | 13
lines Add ActionID to CoreShowChannel event. There is
inconsistency in how we handle manager responses that are lists
of items and, unfortunately, third parties have come to rely on
ActionID being on every event within those lists instead of just
keeping track of the ActionID for the current response. This
change makes CoreShowChannels include the ActionID with each
CoreShowChannel event generated as a result of it being called.
(closes issue #15001) Reported by: sum Patches:
patchactionid2.patch uploaded by sum (license 766) ........
2009-05-26 22:44 +0000 [r196870-196949] Russell Bryant <russell@digium.com>
* /, autoconf/ast_check_osptk.m4 (added), configure,
include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
196946 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r196946 |
russell | 2009-05-26 17:40:34 -0500 (Tue, 26 May 2009) | 8 lines
Update configure script to check for OSP toolkit 3.5.0. (closes
issue #14988) Reported by: tzafrir Patches: configure.ac.diff
uploaded by homesick (license 91) new_ast_check_osptk.m4 uploaded
by homesick (license 91) ........
* /, res/res_convert.c: Merged revisions 196843 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r196843 | russell | 2009-05-26 13:20:57 -0500 (Tue, 26 May 2009)
| 16 lines Merged revisions 196826 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r196826 | russell | 2009-05-26 13:14:36 -0500 (Tue, 26 May 2009)
| 9 lines Resolve a file handle leak. The frames here should have
always been freed. However, out of luck, there was never any
memory leaked. However, after file streams became reference
counted, this code would leak the file stream for the file being
read. (closes issue #15181) Reported by: jkroon ........
................
2009-05-26 16:39 +0000 [r196793] Sean Bright <sean.bright@gmail.com>
* apps/app_queue.c, /: Merged revisions 196792 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r196792 |
seanbright | 2009-05-26 12:38:54 -0400 (Tue, 26 May 2009) | 2
lines Add a missing unref for queues in handle_statechange.
........
2009-05-26 13:47 +0000 [r196661-196724] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 196721 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r196721 |
file | 2009-05-26 10:43:13 -0300 (Tue, 26 May 2009) | 7 lines Fix
a bug where the sip unregister CLI command did not completely
unregister the peer. (closes issue #15118) Reported by: alecdavis
Patches: chan_sip_unregister.diff2.txt uploaded by alecdavis
(license 585) ........
* contrib/scripts/safe_asterisk, /: Merged revisions 196658 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r196658 | file | 2009-05-26 10:06:50 -0300 (Tue,
26 May 2009) | 14 lines Merged revisions 196657 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r196657 | file | 2009-05-26 10:06:09 -0300 (Tue, 26 May 2009) | 7
lines Remove some bash specific stuff from safe_asterisk. (closes
issue #10812) Reported by: paravoid Patches:
safe_asterisk_bashism.diff uploaded by tzafrir (license 46)
........ ................
2009-05-23 05:29 +0000 [r196487] Moises Silva <moises.silva@gmail.com>
* channels/chan_dahdi.c, /: Merged revisions 196456 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r196456 | moy | 2009-05-22 23:27:47 -0500 (Fri, 22 May 2009) | 1
line set MFCR2_CATEGORY just when starting the pbx ........
2009-05-22 21:59 +0000 [r196452] David Vossel <dvossel@digium.com>
* configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions
196416 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r196416 |
dvossel | 2009-05-22 16:09:45 -0500 (Fri, 22 May 2009) | 19 lines
SIP set outbound transport type from Registration In sip.conf the
transport option allows for the configuration of what transport
types (udp, tcp, and tls) a peer will accept, but only the first
type listed was used for outbound connections. This patch changes
this. Now the default transport type is only used until the peer
registers. When registration takes place the transport type is
parsed out of the Contact header. If the Contact header's
transport type is equal to one that the peer supports, the peer's
default transport type for outbound connections is set to match
the Contact header's type. If the Contact header's transport type
is not present, then the peer's default transport type is set to
match the one the peer registered with. When a peer unregisters
or the registration expires, the default transport type for that
peer is reset. (closes issue #12282) Reported by: rjain Patches:
reg_patch_1.diff uploaded by dvossel (license 671) Tested by:
dvossel (closes issue #14727) Reported by: pj Patches:
reg_patch_3.diff uploaded by dvossel (license 671) Tested by: pj,
dvossel Review: https://reviewboard.asterisk.org/r/249/ ........
2009-05-22 19:48 +0000 [r196378] Eliel C. Sardanons <eliels@gmail.com>
* /, apps/app_minivm.c: Merged revisions 196377 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r196377 |
eliel | 2009-05-22 15:38:33 -0400 (Fri, 22 May 2009) | 11 lines
Unregister every registered application by MiniVM. The MinivmMWI
application was not being unregistered on unload and we were not
able to load again the module or reload it. (closes issue #15174)
Reported by: junky Patches: unregister_minivm_mwi.diff uploaded
by junky (license 177) ........
2009-05-22 13:59 +0000 [r196120] Joshua Colp <jcolp@digium.com>
* channels/chan_misdn.c, /: Merged revisions 196117 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r196117 | file | 2009-05-22 10:56:47 -0300 (Fri,
22 May 2009) | 12 lines Merged revisions 196116 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r196116 | file | 2009-05-22 10:54:17 -0300 (Fri, 22 May 2009) | 5
lines Fix a bug where using immediate with mISDN caused a cause
code of 16 to get sent back instead of 1 if the 's' extension did
not exist. (closes issue #12286) Reported by: lmamane ........
................
2009-05-21 19:15 +0000 [r196000] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c, /: Merged revisions 195995 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r195995 | dvossel | 2009-05-21 14:11:49 -0500
(Thu, 21 May 2009) | 20 lines Merged revisions 195991 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r195991 | dvossel | 2009-05-21 14:04:56 -0500 (Thu, 21 May 2009)
| 14 lines Sign problem calculating timestamp for iax frame leads
to no audio on the receiving peer. There are rare cases in which
a frame's delivery timestamp is slightly less than the iax2_pvt's
offset. This causes the pvt's timestamp to be a small negative
number, but since the timestamp value is unsigned it looks like a
huge positive number. This patch checks for this negative case
and sets the ms to zero. A similar check is already done right
below this one in the 'else' statement. (closes issue #15032)
Reported by: guillecabeza Patches: chan_iax2.c.patch_timestamp
uploaded by guillecabeza (license 380) Tested by: guillecabeza
(closes issue #14216) Reported by: Andrey Sofronov ........
................
2009-05-21 15:57 +0000 [r195883] Matthew Nicholson <mnicholson@digium.com>
* main/cdr.c, /, include/asterisk/cdr.h: Merged revisions 195882
via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r195882 | mnicholson | 2009-05-21 10:33:55 -0500
(Thu, 21 May 2009) | 20 lines Merged revisions 195881 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r195881 | mnicholson | 2009-05-21 10:25:50 -0500 (Thu, 21 May
2009) | 13 lines This commit prevents cdr records with
AST_CDR_FLAG_ANSLOCKED and AST_CDR_FLAG_LOCKED from being updated
in certain cases. This is accomplished by adding two functions to
update the answer time and disposition of calls that checks for
the proper lock flags. These functions are used in the
ast_bridge_call() function so that ForkCDR(A) calls are
respected. This patch also modifies the way ast_bridge_call()
chooses the cdr record to base the bridged_cdr on. Previously the
first unlocked cdr record would be chosen, now instead the first
cdr record is chosen and forked cdr records are moved to the
bridge_cdr. This allows the original cdr record and any forked
cdr records to be properly updated with answer and end times.
(closes issue #13797) Reported by: sh0t Tested by: sh0t (closes
issue #14744) Reported by: deepesh ........ ................
2009-05-20 23:31 +0000 [r195842] Tilghman Lesher <tlesher@digium.com>
* apps/app_stack.c, /: Merged revisions 195839 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r195839 |
tilghman | 2009-05-20 18:30:05 -0500 (Wed, 20 May 2009) | 3 lines
If a variable had a blank value upon the initial setting, then it
would do nothing. Identified by Dmitry Andrianov via private
email, fixed by me. ........
2009-05-20 17:35 +0000 [r195639-195707] Joshua Colp <jcolp@digium.com>
* /, main/features.c: Merged revisions 195698 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r195698 | file | 2009-05-20 14:33:02 -0300 (Wed, 20 May 2009) |
12 lines Merged revisions 195688 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r195688 | file | 2009-05-20 14:30:25 -0300 (Wed, 20 May 2009) | 5
lines Fix some code that wrongly assumed a pointer would always
be non-NULL when dealing with CDRs after a bridge. (closes issue
#15079) Reported by: barryf ........ ................
* /, apps/app_meetme.c: Merged revisions 195636 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r195636 | file | 2009-05-20 14:14:42 -0300 (Wed, 20 May 2009) |
12 lines Merged revisions 195635 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r195635 | file | 2009-05-20 14:14:00 -0300 (Wed, 20 May 2009) | 5
lines Fix a bug where the MeetMe option 'D' did not actually
prompt for the pin. (closes issue #15050) Reported by: pmhaddad
........ ................
2009-05-19 20:19 +0000 [r195531] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c, /: Merged revisions 195521 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r195521 | tilghman | 2009-05-19 15:16:01 -0500
(Tue, 19 May 2009) | 14 lines Merged revisions 195520 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r195520 | tilghman | 2009-05-19 15:12:20 -0500 (Tue, 19 May 2009)
| 7 lines Ensure thread keys are initialized before attempting to
access them. (closes issue #14889) Reported by: jaroth Patches:
app_voicemail.c.patch uploaded by msirota (license 758) Tested
by: msirota, BlargMaN ........ ................
2009-05-19 14:49 +0000 [r195452] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 195449 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r195449 | file | 2009-05-19 11:43:54 -0300 (Tue, 19 May 2009) |
14 lines Merged revisions 195448 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r195448 | file | 2009-05-19 11:41:45 -0300 (Tue, 19 May 2009) | 7
lines Fix a bug where direct RTP setup would partially occur even
when disabled if the calling channel was answered. (issue #13545)
Reported by: davidw (issue #14244) Reported by: mbnwa ........
................
2009-05-18 21:25 +0000 [r195405] Eliel C. Sardanons <eliels@gmail.com>
* main/manager.c, /: Merged revisions 195369 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r195369 |
eliel | 2009-05-18 16:49:20 -0400 (Mon, 18 May 2009) | 8 lines
Fix the CLI command 'manager show command' documentation and
functionality. The CLI command 'manager show command' supports
passing multiple action names in the same line, but it was not
allowing that because of a incorrect check in the argumentes
counter. Also the documentation was updated to show that this
usage of the command is possible. ........
2009-05-18 20:55 +0000 [r195359-195373] Tilghman Lesher <tlesher@digium.com>
* apps/app_queue.c, include/asterisk/smdi.h, res/res_monitor.c,
apps/app_voicemail.c, res/res_smdi.c, /,
include/asterisk/monitor.h: Merged revisions 195370 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r195370 | tilghman | 2009-05-18 15:52:33 -0500
(Mon, 18 May 2009) | 15 lines Recorded merge of revisions 195366
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r195366 | tilghman | 2009-05-18 15:24:13 -0500 (Mon, 18 May 2009)
| 8 lines Add a similar dependency on SMDI for voicemail as
already exists for ADSI. (closes issue #14846) Reported by: pj
Patches: 20090413__bug14846__1.4.diff.txt uploaded by tilghman
(license 14) 20090507__issue14846__1.6.0.diff.txt uploaded by
tilghman (license 14) 20090507__issue14846__1.6.1.diff.txt
uploaded by tilghman (license 14) ........ ................
* main/asterisk.c, /: Merged revisions 195320 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r195320 |
tilghman | 2009-05-18 14:17:15 -0500 (Mon, 18 May 2009) | 9 lines
Move the spawn of astcanary down, until after the call to
daemon(3). This avoids possible conflicts with the internal
implementation of daemon(3). (closes issue #15093) Reported by:
tzafrir Patches: 20090513__issue15093__2.diff.txt uploaded by
tilghman (license 14) Tested by: tzafrir ........
2009-05-18 19:01 +0000 [r195319] Mark Michelson <mmichelson@digium.com>
* apps/app_externalivr.c, /: Merged revisions 195316 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r195316 | mmichelson | 2009-05-18 13:58:26 -0500 (Mon, 18 May
2009) | 18 lines Fix externalivr's setvariable command so that it
properly sets multiple variables. The command had a for loop that
was guaranteed to only execute once since the continuation
operation of the loop would set the input buffer NULL. I rewrote
the loop so that its operation was more obvious, and it would set
multiple variables correctly. I also reduced stack space required
for the function, constified the input string, and modified the
function so that it would not modify the input string while I was
at it. (closes issue #15114) Reported by: chris-mac Patches:
15114.patch uploaded by mmichelson (license 60) Tested by:
chris-mac ........
2009-05-18 15:57 +0000 [r195212] Joshua Colp <jcolp@digium.com>
* main/frame.c, /: Merged revisions 195207 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r195207 | file | 2009-05-18 12:53:26 -0300 (Mon, 18 May 2009) |
14 lines Merged revisions 195206 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r195206 | file | 2009-05-18 12:51:22 -0300 (Mon, 18 May 2009) | 7
lines Fix a typo which caused loss of audio when using G729 in
some scenarios with a smoother present. (closes issue #15105)
Reported by: bamby Patches: process-vad-correctly.diff uploaded
by bamby (license 430) ........ ................
2009-05-18 14:54 +0000 [r195164] Eliel C. Sardanons <eliels@gmail.com>
* apps/app_dial.c, main/pbx.c, /, apps/app_macro.c: Merged
revisions 195162 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r195162 |
eliel | 2009-05-18 10:45:23 -0400 (Mon, 18 May 2009) | 9 lines
Warn about the use of the application WaitExten() within a
Macro(). Update applications documentation to warn the user about
the use of the WaitExten() application within a Macro().
Recommend the use of Read() instead. (closes issue #14444)
Reported by: ewieling ........
2009-05-18 14:00 +0000 [r195099] Joshua Colp <jcolp@digium.com>
* main/rtp.c, /: Merged revisions 195096 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r195096 | file | 2009-05-18 10:56:16 -0300 (Mon, 18 May 2009) |
12 lines Merged revisions 195095 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r195095 | file | 2009-05-18 10:53:39 -0300 (Mon, 18 May 2009) | 5
lines Fix a bug where the codecs of the called party leg were not
properly sent back to the caller call leg when reinvited. (closes
issue #13569) Reported by: bkw918 ........ ................
2009-05-18 13:50 +0000 [r195093-195094] Eliel C. Sardanons <eliels@gmail.com>
* /, main/xml.c: Merged revisions 195075 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r195075 |
eliel | 2009-05-18 09:30:34 -0400 (Mon, 18 May 2009) | 3 lines Do
not avoid loading the XML documentation if not XInclude
substitution is done. ........
* doc/appdocsxml.dtd, Makefile, /, main/xml.c: Merged revisions
194982 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r194982 |
eliel | 2009-05-16 16:01:22 -0400 (Sat, 16 May 2009) | 20 lines
Allow to include sections of other parts of the xml
documentation. Avoid duplicating xml documentation by allowing to
include other parts of the xml documentation using XInclude.
Example: <xi:include
xpointer="xpointer(/docs/function[@name='CHANNEL']/synopsis)" />
(Insert this line to include the synopsis of the CHANNEL function
xml documentation). It is also possible to include documentation
from other files in the 'documentation/' directory using the
href="" attribute inside a xinclude element. (closes issue
#15107) Reported by: lmadsen (issue #14444) Reported by: ewieling
........
2009-05-18 13:39 +0000 [r195092] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 195089 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r195089 |
file | 2009-05-18 10:36:17 -0300 (Mon, 18 May 2009) | 5 lines Fix
a bug where specifying an empty outboundproxy would cause packets
to get sent to ourself. (closes issue #15106) Reported by:
timeshell ........
2009-05-18 13:14 +0000 [r195024] Russell Bryant <russell@digium.com>
* main/manager.c, /: Merged revisions 195021 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r195021 | russell | 2009-05-18 07:59:11 -0500 (Mon, 18 May 2009)
| 12 lines Recorded merge of revisions 195020 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r195020 | russell | 2009-05-18 07:57:46 -0500 (Mon, 18 May 2009)
| 5 lines Don't try to unlock a bogus channel. (closes issue
#15144) Reported by: cristiandimache ........ ................
2009-05-16 18:43 +0000 [r194946] Eliel C. Sardanons <eliels@gmail.com>
* main/pbx.c, /: Merged revisions 194945 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r194945 |
eliel | 2009-05-16 14:32:11 -0400 (Sat, 16 May 2009) | 8 lines
Fix a missing unlock in case of error, and a missing free().
Always free the allocated memory for a string field, because we
are always using it (not only when xmldocs are enabled). Also if
there is an error allocating memory for the string field remember
to unlock the list of registered applications, before returning.
........
2009-05-15 22:48 +0000 [r194836-194877] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c, /: Merged revisions 194874 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r194874 | dvossel | 2009-05-15 17:44:44 -0500
(Fri, 15 May 2009) | 23 lines Merged revisions 194873 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r194873 | dvossel | 2009-05-15 17:43:13 -0500 (Fri, 15 May 2009)
| 17 lines IAX2 REGAUTH loop IAX was not sending REGREJ to
terminate invalid registrations. Instead it sent another REGAUTH
if the authentication challenge failed. This caused a loop of
REGREQ and REGAUTH frames. (Related to Security fix AST-2009-001)
(closes issue #14867) Reported by: aragon Tested by: dvossel
(closes issue #14717) Reported by: mobeck Patches:
regauth_loop_update_patch.diff uploaded by dvossel (license 671)
Tested by: dvossel ........ ................
* channels/chan_iax2.c, channels/iax2-parser.c,
channels/iax2-parser.h, /, channels/iax2.h: Merged revisions
194833 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r194833 | dvossel | 2009-05-15 15:52:12 -0500 (Fri, 15 May 2009)
| 24 lines Merged revisions 194557,194685 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r194557 | dvossel | 2009-05-14 17:59:43 -0500 (Thu, 14 May 2009)
| 10 lines IAX2 "Ghost" Channels There is a bug tracker issue
where people are reporting "Ghost" channels in their 'iax2 show
channels' output. The confusion is caused by channels being
listed as "(NONE)" with format "unknown". These are not channels
of coarse. They are usually just pending registration or poke
requests, but it is confusing output. To help make sense of this
I have added two columns to 'iax2 show channels'. One shows the
first message which started the transaction, and the second shows
the last message sent by either side of the call. This helps
diagnose why the entry exists and why it may not go away. (closes
issue #14207) Reported by: clive18 Review:
https://reviewboard.asterisk.org/r/246/ ........ r194685 |
dvossel | 2009-05-15 10:40:37 -0500 (Fri, 15 May 2009) | 6 lines
Update to previous IAX2 "Ghost" Channels patch. Fixed some
comments made on reviewboard for the previous patch. (issue
#14207) ........ ................
2009-05-15 18:44 +0000 [r194717-194768] Russell Bryant <russell@digium.com>
* configs/logger.conf.sample, /: Merged revisions 194765 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r194765 | russell | 2009-05-15 13:43:42 -0500
(Fri, 15 May 2009) | 10 lines Merged revisions 194764 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r194764 | russell | 2009-05-15 13:43:18 -0500 (Fri, 15 May 2009)
| 2 lines Fix some spelling fail. ........ ................
* /, codecs/g722/g722_encode.c, codecs/g722/g722_decode.c: Merged
revisions 194722 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r194722 |
russell | 2009-05-15 12:59:08 -0500 (Fri, 15 May 2009) | 4 lines
Shuttle some bits around to address some gain issues with G.722.
(closes AST-209) ........
* codecs/Makefile, codecs/g722/Makefile (removed), /: Merged
revisions 194718 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r194718 |
russell | 2009-05-15 12:37:12 -0500 (Fri, 15 May 2009) | 2 lines
Further simplify codec_g722 build. ........
* codecs/Makefile, /: Merged revisions 194714 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r194714 |
russell | 2009-05-15 12:24:39 -0500 (Fri, 15 May 2009) | 2 lines
Actually force running make for g722. ........
2009-05-15 13:47 +0000 [r194650] Michiel van Baak <michiel@vanbaak.info>
* CREDITS, /: Merged revisions 194649 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r194649 |
mvanbaak | 2009-05-15 15:43:24 +0200 (Fri, 15 May 2009) | 2 lines
add eliel ........
2009-05-15 13:42 +0000 [r194648] Eliel C. Sardanons <eliels@gmail.com>
* doc/appdocsxml.dtd, main/xmldoc.c, /: Merged revisions 194635 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r194635 | eliel | 2009-05-15 09:23:37 -0400 (Fri, 15 May
2009) | 16 lines Allow to specify an enumlist inside an enum. It
was not possible to use an enumlist inside an enum: <enumlist>
<enum name="aa"> <enumlist> ... </enumlist> </enum> </enumlist>
Now we will be able to insert as many levels as we want. (closes
issue #15112) Reported by: lmadsen ........
2009-05-14 22:31 +0000 [r194545] Kevin P. Fleming <kpfleming@digium.com>
* /: Merged revisions 194520 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r194520 | kpfleming | 2009-05-14 17:26:02 -0500 (Thu, 14 May
2009) | 9 lines Merged revisions 194509 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r194509 | kpfleming | 2009-05-14 17:23:49 -0500 (Thu, 14 May
2009) | 1 line Update URL to Reviewboard ........
................
2009-05-14 22:23 +0000 [r194510] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 194496 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r194496 | mmichelson | 2009-05-14 17:20:51 -0500 (Thu, 14 May
2009) | 30 lines Merged revisions 194484 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r194484 | mmichelson | 2009-05-14 17:17:55 -0500 (Thu, 14 May
2009) | 24 lines Fix a race condition where a reinvite could
trigger a 482 response. The loop detection/spiral detection code
in chan_sip used the owner channel's state as a criterion for
determining if the incoming INVITE is a looped request. The
problem with this is that the INVITE-handling code happens in a
different thread than the thread that marks the owner channel as
being up. As a result, if a reinvite were to come in very
quickly, say from another Asterisk on the same LAN, it was
possible for the reinvite to arrive before the owner channel had
been set to the up state. This patch corrects the problem by
using the invitestate of the sip_pvt instead, since that can be
guaranteed to be set correctly by the time the reinvite arrives.
Since there is a switch statement further in the INVITE-handling
code, the AST_STATE_RINGING state also checks the invitestate of
the sip_pvt in case we should actually be treating the channel as
if it were up already. (closes issue #12215) Reported by: jpyle
Patches: 12215_confirmed.patch uploaded by mmichelson (license
60) Tested by: lmadsen ........ ................
2009-05-14 17:07 +0000 [r194437] Joshua Colp <jcolp@digium.com>
* /, apps/app_meetme.c: Merged revisions 194434 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r194434 |
file | 2009-05-14 14:05:33 -0300 (Thu, 14 May 2009) | 7 lines Fix
a bug where the 'T' option to Meetme did not work. (closes issue
#15031) Reported by: Stochastic (closes issue #13801) Reported
by: justdave ........
2009-05-14 16:23 +0000 [r194431] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c, /: Merged revisions 194430 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r194430 |
tilghman | 2009-05-14 11:22:14 -0500 (Thu, 14 May 2009) | 7 lines
If the timing ended on a zero, then we would loop forever.
(closes issue #14983) Reported by: teox Patches:
20090513__issue14983.diff.txt uploaded by tilghman (license 14)
Tested by: teox ........
2009-05-13 13:42 +0000 [r194213] Joshua Colp <jcolp@digium.com>
* main/rtp.c, /: Merged revisions 194209 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r194209 | file | 2009-05-13 10:39:10 -0300 (Wed, 13 May 2009) |
18 lines Merged revisions 194208 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r194208 | file | 2009-05-13 10:38:01 -0300 (Wed, 13 May 2009) |
11 lines Fix RFC2833 issues with DTMF getting duplicated and with
duration wrapping over. (closes issue #14815) Reported by:
geoff2010 Patches: v1-14815.patch uploaded by dimas (license 88)
Tested by: geoff2010, file, dimas, ZX81, moliveras (closes issue
#14460) Reported by: moliveras Tested by: moliveras ........
................
2009-05-13 00:54 +0000 [r194141] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c, /: Merged revisions 194138 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r194138 | tilghman | 2009-05-12 19:52:49 -0500 (Tue, 12 May 2009)
| 14 lines Merged revisions 194137 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r194137 | tilghman | 2009-05-12 19:52:03 -0500 (Tue, 12 May 2009)
| 7 lines Fix logic for how to proceed with a single digit
extension. (closes issue #15091) Reported by: andrew Patches:
20090512__issue15091.diff.txt uploaded by tilghman (license 14)
Tested by: andrew ........ ................
2009-05-12 22:48 +0000 [r194059] Matthew Nicholson <mnicholson@digium.com>
* apps/app_queue.c, /: Merged revisions 194057 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r194057 | mnicholson | 2009-05-12 17:32:13 -0500 (Tue, 12 May
2009) | 22 lines Merged revisions 194028 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r194028 | mnicholson | 2009-05-12 17:15:45 -0500 (Tue, 12 May
2009) | 16 lines This change modifies app_queue to properly
generate CDR records in failure situations. This involves setting
a proper cdr disposition coresponding to the given failure
condition and ensuring the proper information is stored in the
cdr record. (closes issue #13691) Reported by: dferrer Tested by:
mnicholson (closes issue #13637) Reported by: atis Tested by:
atis ........ ................
2009-05-12 20:51 +0000 [r193962] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 193954 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r193954 |
mmichelson | 2009-05-12 15:28:13 -0500 (Tue, 12 May 2009) | 18
lines Update spiral support in trunk and 1.6.X to match what is
in 1.4. In 1.4, a SIP spiral is treated the same way as a call
forward. This works much better than what is currently in trunk
and 1.6.X. The code in trunk and 1.6.X did not create a new call
to the recipient of the spiral, instead trying to continue the
same call. In addition to just being plain wrong, this also had
the side effect of only being able to spiral calls to other SIP
channels. With this in place, as long as call forwards are
honored, SIP spirals will work properly. This means that it will
work for outbound calls made by the Queue, Dial, and Page
applications. For originated calls and spool calls, however, the
spiral will not work properly until a generic call forward
mechanism is introduced into Asterisk. (relates to issue #13630)
........
2009-05-12 20:42 +0000 [r193823-193959] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c, /: Merged revisions 193956 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r193956 | tilghman | 2009-05-12 15:40:22 -0500
(Tue, 12 May 2009) | 13 lines Merged revisions 193955 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r193955 | tilghman | 2009-05-12 15:39:21 -0500 (Tue, 12 May 2009)
| 6 lines Avoid initializing routines if the authentication
fails. Fixes a crash (RR) issue. (closes issue #14508) Reported
by: tiziano Patches: 20090221_2_wrongmailbox.diff.txt uploaded by
tiziano (license 377) ........ ................
* apps/app_voicemail.c, /: Merged revisions 193870 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r193870 | tilghman | 2009-05-12 12:29:33 -0500 (Tue, 12 May 2009)
| 2 lines Convert a THREADSTORAGE object into a simple malloc'd
object (as suggested by Russell on -dev) ........
* apps/app_voicemail.c, /: Recorded merge of revisions 193756 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r193756 | tilghman | 2009-05-11 17:50:47 -0500
(Mon, 11 May 2009) | 25 lines Recorded merge of revisions 193755
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r193755 | tilghman | 2009-05-11 17:48:20 -0500 (Mon, 11 May 2009)
| 18 lines Move 300 bytes around on the stack, to make more room
for an extension buffer. This allows more concurrent extensions
to be copied for a single voicemail, without creating a
possibility of upsetting existing users, where a dialplan could
run out of stack space where it had run fine before.
Alternatively, we could have allocated off the heap, but that is
a larger change and would have increased the chance for
instability introduced by this change. This is really solved
starting in 1.6.0.11, as the use of an ast_str buffer allows an
unlimited number of extensions (up to available memory). We
additionally create a new warning message when the buffer length
is exceeded, permitting administrators to see an issue after the
fact, whereas previously the list was silently truncated. (closes
issue #14739) Reported by: p_lindheimer Patches:
20090417__bug14739.diff.txt uploaded by tilghman (license 14)
Tested by: p_lindheimer ........ ................
2009-05-11 22:12 +0000 [r193719] Russell Bryant <russell@digium.com>
* /, res/res_timing_timerfd.c: Merged revisions 193718 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r193718 | russell | 2009-05-11 17:04:40 -0500 (Mon, 11 May 2009)
| 12 lines Fix some timer state corruption. In res_timer_timerfd,
handle the case that set_rate gets called while a timer is still
in continuous mode. In this case, we want to remember the
configured rate, but not actually set it until continuous mode
has been disabled. Thanks to dvossel for finding and helping to
debug the problem. (closes issue #15080) Reported by: dvossel
Tested by: dvossel ........
2009-05-11 19:17 +0000 [r193617] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c, /: Merged revisions 193614 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r193614 | rmudgett | 2009-05-11 14:11:29 -0500
(Mon, 11 May 2009) | 19 lines Merged revisions 193613 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r193613 | rmudgett | 2009-05-11 14:09:00 -0500 (Mon, 11 May 2009)
| 12 lines Sent wrong message to clear a call we started if the
other end has not responed yet. In the state MISDN_CALLING (i.e.
SETUP was sent but no answer has arrived yet), it is not allowed
to clear the call with RELEASE_COMPLETE. It must be cleared with
DISCONNECT. A RELEASE_COMPLETE is only allowed as an answer to a
SETUP. (See Q.931 ch. 5.3.2, 5.3.2.a, 5.3.2.b) Patches:
chan-misdn-ccstate7.patch uploaded by customer. JIRA ABE-1862
........ ................
2009-05-11 18:59 +0000 [r193612] Leif Madsen <lmadsen@digium.com>
* /, funcs/func_channel.c: Update CHANNEL(transfercapabilities)
documentation. (closes issue #15073) Reported by: pkempgen
Patches: 20090511__issue15073__trunk.diff.txt uploaded by
tilghman (license 14)
2009-05-10 17:08 +0000 [r193503] Joshua Colp <jcolp@digium.com>
* main/bridging.c, /: Merged revisions 193502 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r193502 |
file | 2009-05-10 14:07:46 -0300 (Sun, 10 May 2009) | 2 lines Fix
a bug where receiving a control frame of subclass -1 would cause
certain channels to get hung up. ........
2009-05-09 11:33 +0000 [r193462] Russell Bryant <russell@digium.com>
* include/asterisk/event.h, /: Merged revisions 193461 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r193461 | russell | 2009-05-09 06:33:09 -0500 (Sat, 09 May 2009)
| 2 lines Minor documentation update for ast_event_queue().
........
2009-05-08 20:52 +0000 [r193390] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 193387 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r193387 |
dvossel | 2009-05-08 15:32:51 -0500 (Fri, 08 May 2009) | 7 lines
TCP not matching valid peer. find_peer() does not find a valid
peer when using pvt->recv as the sockaddr_in argument. Because of
the way TCP works, the port number in pvt->recv is not what we're
looking for at all. There is currently only one place that
find_peer searches for a peer using the sockaddr_in argument. If
the peer is not found after using pvt->recv (works for UDP since
the port number will be correct), a temp sockaddr_in struct is
made using the Contact header in the sip_request. This has the
correct port number in it. Review:
http://reviewboard.digium.com/r/236/ ........
2009-05-08 19:51 +0000 [r193350] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c, /: Merged revisions 193349 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r193349 |
mmichelson | 2009-05-08 14:50:44 -0500 (Fri, 08 May 2009) | 12
lines Reset the members' call counts when resetting queue
statistics. This helps to prevent odd scenarios where a queue
will claim to have taken 0 calls, but the members appear to have
taken a non-zero amount. (closes issue #15068) Reported by: sum
Patches: patchreset.patch uploaded by sum (license 766) Tested
by: sum ........
2009-05-08 15:36 +0000 [r193336] Sean Bright <sean.bright@gmail.com>
* funcs/func_devstate.c, /: Merged revisions 193274 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r193274 | seanbright | 2009-05-08 11:18:40 -0400 (Fri, 08 May
2009) | 2 lines Fix the spelling of UNAVAILABLE in func_devstate
CLI completion. ........
2009-05-08 14:55 +0000 [r193266] David Vossel <dvossel@digium.com>
* channels/misdn_config.c, /: Merged revisions 193263 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r193263 | dvossel | 2009-05-08 09:52:19 -0500
(Fri, 08 May 2009) | 15 lines Merged revisions 193262 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r193262 | dvossel | 2009-05-08 09:51:09 -0500 (Fri, 08 May 2009)
| 9 lines "misdn show config" segfaults asterisk, if no MSN lists
(closes issue #14976) Reported by: alecdavis Patches:
misdn_config.diff.txt uploaded by alecdavis (license 585) Tested
by: alecdavis, FabienToune ........ ................
2009-05-08 14:12 +0000 [r193197] Kevin P. Fleming <kpfleming@digium.com>
* configs/logger.conf.sample, /, main/logger.c: Merged revisions
193194 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r193194 | kpfleming | 2009-05-08 09:06:15 -0500 (Fri, 08 May
2009) | 13 lines Merged revisions 193193 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r193193 | kpfleming | 2009-05-08 09:03:28 -0500 (Fri, 08 May
2009) | 7 lines Make absolute paths for logger channels work
properly (Note: This is not a new feature, it was previously
undocumented and broken.) The Asterisk logger has a feature to
support absolute pathnames for logger channels, but the code
implementing the feature was broken. This has been fixed, and the
absolute path feature is now documented in the sample
logger.conf. ........ ................
2009-05-07 23:44 +0000 [r193123] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c, /: Merged revisions 193120 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r193120 | tilghman | 2009-05-07 18:42:28 -0500 (Thu, 07 May 2009)
| 26 lines Merged revisions 193119 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r193119 | tilghman | 2009-05-07 18:41:11 -0500 (Thu, 07 May 2009)
| 19 lines Fix Background within a Macro for FreePBX. If the
single digit DTMF is an extension in the specified context, then
go there and signal no DTMF. Otherwise, we should exit with that
DTMF. If we're in Macro, we'll exit and seek that DTMF as the
beginning of an extension in the Macro's calling context. If
we're not in Macro, then we'll simply seek that extension in the
calling context. Previously, someone complained about the
behavior as it related to the interior of a Gosub routine, and
the fix (#14011) inadvertently broke FreePBX (#14940). This
change should fix both of these situations, but with the possible
incompatibility that if a single digit extension does not exist
(but a longer extension COULD have matched), it would have
previously gone immediately to the "i" extension, but will now
need to wait for a timeout. (closes issue #14940) Reported by:
p_lindheimer Patches: 20090420__bug14940.diff.txt uploaded by
tilghman (license 14) Tested by: p_lindheimer ........
................
2009-05-07 22:51 +0000 [r193080] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c, /: Merged revisions 193077 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r193077 | rmudgett | 2009-05-07 17:24:04 -0500
(Thu, 07 May 2009) | 12 lines Merged revisions 193050 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r193050 | rmudgett | 2009-05-07 17:17:06 -0500 (Thu, 07 May 2009)
| 5 lines Give a more helpful message when an incoming call's
dialed extension does not match. Added the dialed extension and
context to the chan_misdn messages warning that the dialed number
cannot be matched in the dialplan. ........ ................
2009-05-07 17:53 +0000 [r192936-193008] Tilghman Lesher <tlesher@digium.com>
* /, funcs/func_odbc.c: Merged revisions 193006 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r193006 |
tilghman | 2009-05-07 12:51:13 -0500 (Thu, 07 May 2009) | 7 lines
Second result should not contain data from the first result.
(closes issue #15039) Reported by: jims Patches:
20090506__issue15039.diff.txt uploaded by tilghman (license 14)
Tested by: jims ........
* channels/chan_unistim.c, /: Merged revisions 192938 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r192938 | tilghman | 2009-05-07 12:13:36 -0500 (Thu, 07 May 2009)
| 6 lines Send DTMF frame before playing back audio. (closes
issue #14858) Reported by: barryf Patches:
20090507__bug14858.diff.txt uploaded by tilghman (license 14)
........
* /, channels/chan_sip.c: Merged revisions 192933 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r192933 | tilghman | 2009-05-07 11:43:56 -0500 (Thu, 07 May 2009)
| 17 lines Merged revisions 192932 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r192932 | tilghman | 2009-05-07 11:29:08 -0500 (Thu, 07 May 2009)
| 10 lines Eliminate repetition of fullcontact during
reconstruction. If the fullcontact field appears in both the
sippeers and the sipregs table, then during reconstruction of the
field, it will otherwise be doubled. (closes issue #14754)
Reported by: Alexei Gradinari Patches:
20090506__bug14754.diff.txt uploaded by tilghman (license 14)
Tested by: lmadsen ........ ................
2009-05-07 Leif Madsen <lmadsen@digium.com>
* Release Asterisk 1.6.2.0-beta2
2009-05-06 22:20 +0000 [r192874] Jeff Peeler <jpeeler@digium.com>
* /, main/features.c: Merged revisions 192861 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r192861 | jpeeler | 2009-05-06 17:17:27 -0500 (Wed, 06 May 2009)
| 17 lines Merged revisions 192858 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r192858 | jpeeler | 2009-05-06 17:15:19 -0500 (Wed, 06 May 2009)
| 10 lines Make ParkedCall application stop execution of the
dialplan after hang up Just changed park_exec to always return
non-zero. I really wasn't entirely sure at first if this was a
bug. Decided it was since it would be surprising when not using
ParkedCall in the dialplan to hang up and have dialplan execution
continue. (closes issue #14555) Reported by: francesco_r ........
................
2009-05-06 17:57 +0000 [r192813] Matthew Fredrickson <creslin@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 190946 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r190946 | mattf | 2009-04-28 17:05:05 -0500 (Tue, 28 Apr 2009) |
1 line Make sure that we do not clear the down flag on the BRI
during PTMP link transients. Also refix SS7 audio that the early
media patch broke. ........
2009-05-06 17:41 +0000 [r192637-192810] Joshua Colp <jcolp@digium.com>
* channels/chan_iax2.c, /: Merged revisions 192808 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r192808 | file | 2009-05-06 14:38:51 -0300 (Wed, 06 May 2009) |
10 lines Fix a bug where a timer would be created but not
acknowledged. This scenario crept up if chan_iax2 was loaded with
no configuration file present. It would create a timer and tell
it to go at an interval but the thread that normally acknowledges
it would not be created because no configuration file was
present. The timer will now be closed if no configuration file is
present. (closes issue #15014) Reported by: madkins ........
* res/res_clialiases.c, /: Merged revisions 192736 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r192736 | file | 2009-05-06 13:09:27 -0300 (Wed, 06 May 2009) | 4
lines Make the code that prevents an infinite loop from happening
into a case insensitive check. (thanks eliel) ........
* res/res_clialiases.c, /: Merged revisions 192700 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r192700 | file | 2009-05-06 11:35:47 -0300 (Wed, 06 May 2009) | 5
lines Fix an infinite loop with tab completion of CLI aliases
that reference themselves. (closes issue #15020) Reported by:
junky ........
* /, channels/chan_sip.c: Merged revisions 192634 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r192634 | file | 2009-05-06 10:34:35 -0300 (Wed, 06 May 2009) |
14 lines Merged revisions 192633 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r192633 | file | 2009-05-06 10:30:51 -0300 (Wed, 06 May 2009) | 7
lines Update some old logic to stop both begin and end DTMF
frames from reaching the core if rfc2833 is not enabled. (closes
issue #15036) Reported by: dimas Patches: v1-15036.patch uploaded
by dimas (license 88) ........ ................
2009-05-05 20:02 +0000 [r192528] Sean Bright <sean.bright@gmail.com>
* /, static-http/astman.js: Merged revisions 192525 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r192525 | seanbright | 2009-05-05 15:57:49 -0400
(Tue, 05 May 2009) | 18 lines Merged revisions 192524 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r192524 | seanbright | 2009-05-05 15:56:11 -0400 (Tue, 05 May
2009) | 11 lines Fix Javascript error when using astman.js in
Internet Explorer. Internet Explorer (tested with 7.0) does not
like trailing commas on constructs like object initializers, so
get rid of them to avoid some errors. (closes issue #15026)
Reported by: rajnishgiri Patches: bug15026.patch uploaded by
seanbright (license 71) Tested by: seanbright ........
................
2009-05-05 18:27 +0000 [r192402-192480] Joshua Colp <jcolp@digium.com>
* /, main/features.c: Merged revisions 192462 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r192462 | file | 2009-05-05 15:23:58 -0300 (Tue, 05 May 2009) |
15 lines Merged revisions 192454 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r192454 | file | 2009-05-05 15:22:27 -0300 (Tue, 05 May 2009) | 8
lines Fix an incorrect assumption that certain values on the
channel will always exist when they may not. The CDR code
involved with bridges wrongly assumed that the currently
executing application and data values will always exist. It is
possible for this to be false when call forwarding is involved.
(closes issue #14984) Reported by: gincantalupo ........
................
* apps/app_followme.c, /: Merged revisions 192430 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r192430 | file | 2009-05-05 14:46:51 -0300 (Tue, 05 May 2009) |
12 lines Merged revisions 192429 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r192429 | file | 2009-05-05 14:43:30 -0300 (Tue, 05 May 2009) | 5
lines Fix a bug where the followme application would continue
trying numbers after the caller hung up. (closes issue #13624)
Reported by: sgenyuk ........ ................
* /, channels/chan_sip.c: Merged revisions 192387 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r192387 |
file | 2009-05-05 11:22:47 -0300 (Tue, 05 May 2009) | 10 lines
Fix a bug with setting t38pt_udptl at the user or peer level. If
an incoming call authenticated as a user or peer and t38pt_udptl
was not set to yes in general then no UDPTL session would be
present and any T38 related things would fail. This commit
changes it so that if after authenticating T38 is enabled but no
UDPTL session is present one will be created. (issue AST-215)
........
2009-05-05 13:43 +0000 [r192298-192360] Kevin P. Fleming <kpfleming@digium.com>
* main/astobj2.c, include/asterisk/stringfields.h, /, main/utils.c:
Merged revisions 192357 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r192357 |
kpfleming | 2009-05-05 15:18:21 +0200 (Tue, 05 May 2009) | 5
lines Correct some flaws in the memory accounting code for
stringfields and ao2 objects Under some conditions, the memory
allocation for stringfields and ao2 objects would not have
supplied valid file/function names for MALLOC_DEBUG tracking, so
this commit corrects that. ........
* main/astobj2.c, main/datastore.c, main/channel.c, /,
include/asterisk/astobj2.h, include/asterisk/datastore.h,
include/asterisk/channel.h: Merged revisions 192318 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r192318 | kpfleming | 2009-05-05 12:34:19 +0200 (Tue, 05 May
2009) | 5 lines Properly account for memory allocated for
channels and datastores As in previous commits, when channels are
allocated (with ast_channel_alloc) or datastores are allocated
(with ast_datastore_alloc) properly account for the memory being
owned by the caller, instead of the allocator function itself.
........
* include/asterisk/stringfields.h, /, main/utils.c: Merged
revisions 192279 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r192279 |
kpfleming | 2009-05-05 10:51:06 +0200 (Tue, 05 May 2009) | 5
lines Ensure that string pools allocated to hold stringfields are
properly accounted in MALLOC_DEBUG mode This commit modifies the
stringfield pool allocator to remember the 'owner' of the
stringfield manager the pool is being allocated for, and ensures
that pools allocated in the future when fields are populated are
owned by that file/function. ........
2009-05-04 22:48 +0000 [r192217] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c, /: Merged revisions 192214 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r192214 | dvossel | 2009-05-04 17:44:51 -0500
(Mon, 04 May 2009) | 17 lines Merged revisions 192213 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r192213 | dvossel | 2009-05-04 17:37:31 -0500 (Mon, 04 May 2009)
| 11 lines global mohinterpret setting is ignored mohinterpret
and mohsuggest global variables were not copied over during
build_users and build_peers. (closes issue #14728) Reported by:
dimas Patches: v1-14728.patch uploaded by dimas (license 88)
Tested by: dimas, dvossel ........ ................
2009-05-04 19:34 +0000 [r192175] Kevin P. Fleming <kpfleming@digium.com>
* main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions
192059 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r192059 |
kpfleming | 2009-05-04 18:24:16 +0200 (Mon, 04 May 2009) | 5
lines Ensure that astobj2 memory allocations are properly
accounted for when MALLOC_DEBUG is used This commit ensures that
all astobj2 allocated objects are properly accounted for in
MALLOC_DEBUG mode by passing down the file/function/line
information from the module/function that actually called the
astobj2 allocation function. ........
2009-05-04 19:31 +0000 [r192135-192173] Tilghman Lesher <tlesher@digium.com>
* /, configure, res/res_agi.c: Merged revisions 192171 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r192171 | tilghman | 2009-05-04 14:29:13 -0500 (Mon, 04 May 2009)
| 8 lines Restore 'asyncagi break' command to 1.6.1 and higher.
(closes issue #14985) Reported by: nikkk Patches:
20090428__bug14985.diff.txt uploaded by tilghman (license 14)
20090429__bug14985__1.6.1.diff.txt uploaded by tilghman (license
14) Tested by: nikkk ........
* autoconf/ast_ext_tool_check.m4, /: Merged revisions 192132 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r192132 | tilghman | 2009-05-04 13:42:56 -0500 (Mon, 04
May 2009) | 6 lines Pass libraries in LIBS, not LDFLAGS. (closes
issue #14671) Reported by: Chainsaw Patches:
asterisk-1.6.0.6-toolcheck-libs-not-ldflags.patch uploaded by
Chainsaw (license 723) ........
2009-05-04 17:45 +0000 [r192097] Leif Madsen <lmadsen@digium.com>
* apps/app_forkcdr.c, /: Merged revisions 192096 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r192096 |
lmadsen | 2009-05-04 13:42:56 -0400 (Mon, 04 May 2009) | 4 lines
Commit documentation changes related to issue #14801. (issue
#14801) ........
2009-05-04 15:54 +0000 [r192033] Eliel C. Sardanons <eliels@gmail.com>
* /, main/xml.c: Merged revisions 192032 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r192032 |
eliel | 2009-05-04 11:35:35 -0400 (Mon, 04 May 2009) | 3 lines Do
not re-define _POSIX_C_SOURCE if it was already defined. ........
2009-05-04 10:01 +0000 [r191958] Kevin P. Fleming <kpfleming@digium.com>
* /, configs/modules.conf.sample: Merged revisions 191955 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r191955 | kpfleming | 2009-05-04 11:57:36 +0200 (Mon, 04
May 2009) | 8 lines Ensure that by default only one console
channel driver is loaded This configuration file was changed to
ensure that only one console channel driver (chan_oss) is loaded
by default, but the change would only work if chan_console was
not built. Now it will work as expected; if chan_alsa or
chan_console are built and installed, they will not be loaded
unless explicity requested. ........
2009-05-03 14:06 +0000 [r191885] Russell Bryant <russell@digium.com>
* Makefile, /: Merged revisions 191884 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r191884 |
russell | 2009-05-03 09:05:10 -0500 (Sun, 03 May 2009) | 2 lines
Remove unnecessary compiler flag ........
2009-05-02 18:48 +0000 [r191779] Kevin P. Fleming <kpfleming@digium.com>
* /, main/logger.c: Merged revisions 191775 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r191775 |
kpfleming | 2009-05-02 20:39:48 +0200 (Sat, 02 May 2009) | 5
lines Fix an error in queue_log file rotation optimization code
This code was copy-and-pasted without properly changing
references to event_rotate into queue_rotate, so under some
conditions the log rotation would rotate queue_log even though it
was not necessary. ........
2009-05-02 15:52 +0000 [r191703] Sean Bright <sean.bright@gmail.com>
* main/asterisk.c, /: Merged revisions 191700 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r191700 |
seanbright | 2009-05-02 11:45:07 -0400 (Sat, 02 May 2009) | 1
line Update copyright year to 2009 ........
2009-05-01 20:02 +0000 [r191554-191563] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_sip.c: Merged revisions 191560 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r191560 | tilghman | 2009-05-01 15:01:21 -0500 (Fri, 01 May 2009)
| 13 lines Merged revisions 191559 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r191559 | tilghman | 2009-05-01 15:00:23 -0500 (Fri, 01 May 2009)
| 6 lines SIP Response 410 maps to cause code 22 (or 23), not 1.
(closes issue #14993) Reported by: BigJimmy Patches: causepatch
uploaded by BigJimmy (license 371) ........ ................
* channels/chan_iax2.c, /: Merged revisions 191494 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r191494 | tilghman | 2009-05-01 13:18:00 -0500 (Fri, 01 May 2009)
| 4 lines Set debug message back to DEBUG level. (closes issue
#15007) Reported by: hulber ........
2009-05-01 18:20 +0000 [r191508] Jeff Peeler <jpeeler@digium.com>
* main/channel.c, /: Merged revisions 191489 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r191489 | jpeeler | 2009-05-01 13:09:23 -0500 (Fri, 01 May 2009)
| 15 lines Merged revisions 191488 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r191488 | jpeeler | 2009-05-01 12:40:46 -0500 (Fri, 01 May 2009)
| 9 lines Fix DTMF not being sent to other side after a partial
feature match This fixes a regression from commit 176701. The
issue was that ast_generic_bridge never exited after the feature
digit timeout had elapsed, which prevented the queued DTMF from
being sent to the other side. This issue was reported to me
directly. ........ ................
2009-04-30 17:46 +0000 [r191224-191370] Tilghman Lesher <tlesher@digium.com>
* main/asterisk.c, /, configure, include/asterisk/autoconfig.h.in,
configure.ac: Merged revisions 191367 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r191367 |
tilghman | 2009-04-30 12:40:58 -0500 (Thu, 30 Apr 2009) | 3 lines
Detect eaccess (or euidaccess) before using it. Reported by
Andrew Lindh via the -dev list. ........
* main/asterisk.c, /: Merged revisions 191283 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r191283 |
tilghman | 2009-04-30 01:47:13 -0500 (Thu, 30 Apr 2009) | 11
lines Change working directory to / under certain conditions. If
backgrounding and no core will be produced, then changing the
directory won't break anything; likewise, if the CWD isn't
accessible by the current user, then a core wasn't possible
anyway. (closes issue #14831) Reported by: chris-mac Patches:
20090428__bug14831.diff.txt uploaded by tilghman (license 14)
20090430__bug14831.diff.txt uploaded by tilghman (license 14)
Tested by: chris-mac ........
* /, channels/h323/ast_h323.cxx, channels/chan_h323.c: Merged
revisions 191219 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r191219 |
tilghman | 2009-04-29 18:06:56 -0500 (Wed, 29 Apr 2009) | 2 lines
Make H.323 compile with FDLEAK detection code enabled ........
2009-04-29 18:40 +0000 [r191139] David Brooks <dbrooks@digium.com>
* pbx/pbx_config.c, /: Merged revisions 191136 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r191136 |
dbrooks | 2009-04-29 13:32:58 -0500 (Wed, 29 Apr 2009) | 3 lines
Removing crufty code that is no longer necessary. Code cleanup.
........
2009-04-29 08:59 +0000 [r190994] Russell Bryant <russell@digium.com>
* main/indications.c, /: Merged revisions 190993 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r190993 |
russell | 2009-04-29 03:58:39 -0500 (Wed, 29 Apr 2009) | 7 lines
Log an error message if indications.conf is not found. (closes
issue #14990) Reported by: tzafrir Patches: indications_err.diff
uploaded by tzafrir (license 46) ........
2009-04-29 06:38 +0000 [r190985] TransNexus OSP Development <support@transnexus.com>
* apps/app_osplookup.c, /: Merged revisions 190830 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r190830 | transnexus | 2009-04-28 17:10:42 +0800 (Tue, 28 Apr
2009) | 2 lines Updated for OSP Toolkit 3.5. ........
2009-04-28 17:33 +0000 [r190907] Tilghman Lesher <tlesher@digium.com>
* doc/tex/cdrdriver.tex, /: Merged revisions 190904 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r190904 | tilghman | 2009-04-28 12:31:43 -0500 (Tue, 28 Apr 2009)
| 2 lines UniqueID column has a maximum size of 150 ........
2009-04-28 14:17 +0000 [r190732-190869] Kevin P. Fleming <kpfleming@digium.com>
* Makefile, /: Merged revisions 190865 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r190865 |
kpfleming | 2009-04-28 09:15:47 -0500 (Tue, 28 Apr 2009) | 5
lines Build XML documention from *only* the source files that
have docs in them Change the build process so that
doc/core-en_US.xml is dependent solely on the source files that
have documentation in them, not on all source files. ........
* /, Makefile.rules: Merged revisions 190861 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r190861 |
kpfleming | 2009-04-28 09:12:09 -0500 (Tue, 28 Apr 2009) | 5
lines Remove Makefile rules for bison and flex sources We never,
ever want these files to processed automatically, because we
store the output files in Subversion and users should never need
to rebuild them. ........
* /, configure, include/asterisk/autoconfig.h.in: Merged revisions
190725 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r190725 | kpfleming | 2009-04-27 14:30:54 -0500 (Mon, 27 Apr
2009) | 13 lines Merged revisions 190721 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r190721 | kpfleming | 2009-04-27 14:29:46 -0500 (Mon, 27 Apr
2009) | 7 lines Fix 'inconsistent line endings' when autoconf
2.63 is used Attempt to make configure script regeneration 'safe'
using autoconf 2.63, which embeds a bare CR into the script, thus
making Subversion complain about inconsistent line endings This
commit changes the MIME type of the configure script to be
'binary' thus making Subversion no longer inspect line endings,
and as a bonus 'svn diff' will no longer try to generate diff
output for it, which is not generally useful anyway. ........
................
2009-04-27 19:36 +0000 [r190729] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c, /: Merged revisions 190726 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r190726 |
tilghman | 2009-04-27 14:34:48 -0500 (Mon, 27 Apr 2009) | 4 lines
Don't warn on pipe in the System call. (closes issue #14979)
Reported by: pj ........
2009-04-27 19:15 +0000 [r190666] Russell Bryant <russell@digium.com>
* res/res_smdi.c, /: Merged revisions 190663 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r190663 | russell | 2009-04-27 14:08:12 -0500 (Mon, 27 Apr 2009)
| 22 lines Merged revisions 190661-190662 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r190661 | russell | 2009-04-27 14:00:54 -0500 (Mon, 27 Apr 2009)
| 9 lines Resolve a crash in res_smdi when used with chan_dahdi.
When chan_dahdi goes to get an SMDI message, it provides no
search criteria. It just grabs the next message that arrives.
This code was written with the SMDI dialplan functions in mind,
since that is now the preferred method of using SMDI. However,
this broke support of it being used from chan_dahdi. (closes
AST-212) ........ r190662 | russell | 2009-04-27 14:03:59 -0500
(Mon, 27 Apr 2009) | 2 lines Fix a typo from 190661. ........
................
2009-04-27 16:28 +0000 [r190625] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c, /: Merged revisions 190622 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r190622 |
mmichelson | 2009-04-27 11:26:14 -0500 (Mon, 27 Apr 2009) | 3
lines Update warning message to not have pipes and contain all
options. ........
2009-04-23 21:23 +0000 [r190383] Russell Bryant <russell@digium.com>
* /, channels/chan_sip.c: Merged revisions 190371 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ ........
2009-04-23 20:44 +0000 [r190355] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c, /: Merged revisions 190352 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r190352 |
tilghman | 2009-04-23 15:42:11 -0500 (Thu, 23 Apr 2009) | 7 lines
Labels are sometimes (most of the time?) NULL for extensions.
(closes issue #14895) Reported by: chris-mac Patches:
20090423__bug14895__2.diff.txt uploaded by tilghman (license 14)
Tested by: lmadsen ........
2009-04-23 19:18 +0000 [r190297] Joshua Colp <jcolp@digium.com>
* channels/chan_local.c, /: Merged revisions 190287 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r190287 | file | 2009-04-23 16:15:30 -0300 (Thu,
23 Apr 2009) | 13 lines Merged revisions 190286 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r190286 | file | 2009-04-23 16:13:18 -0300 (Thu, 23 Apr 2009) | 6
lines Fix a bug in chan_local glare hangup detection. If both
sides of a Local channel were hung up at around the same time it
was possible for one thread to destroy the local private
structure and have the other thread immediately try to remove the
already freed structure from the local channel list. ........
................
2009-04-23 17:47 +0000 [r190253] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c, /: Merged revisions 190250 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r190250 |
mmichelson | 2009-04-23 12:45:35 -0500 (Thu, 23 Apr 2009) | 9
lines Fix reversed behavior of leavewhenempty option in
queues.conf. (closes issue #14650) Reported by: alecdavis
Patches: 14650.patch uploaded by mmichelson (license 60) Tested
by: mmichelson, lmadsen ........
2009-04-22 21:43 +0000 [r190096] Tilghman Lesher <tlesher@digium.com>
* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
include/asterisk/lock.h: Merged revisions 190093 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r190093 | tilghman | 2009-04-22 16:38:15 -0500
(Wed, 22 Apr 2009) | 14 lines Merged revisions 190092 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r190092 | tilghman | 2009-04-22 16:35:03 -0500 (Wed, 22 Apr 2009)
| 7 lines Detect availability of pthread_rwlock_timedwrlock()
before using it. (closes issue #14930) Reported by: tilghman
Patches: 20090420__bug14930.diff.txt uploaded by tilghman
(license 14) Tested by: mvanbaak, tilghman ........
................
2009-04-22 21:18 +0000 [r189997-190066] Jeff Peeler <jpeeler@digium.com>
* main/cli.c, funcs/func_groupcount.c, /, main/app.c,
include/asterisk/channel.h: Merged revisions 190057 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r190057 | jpeeler | 2009-04-22 16:15:55 -0500 (Wed, 22 Apr 2009)
| 9 lines Fix building of chan_h323 with gcc-3.3 There seems to
be a bug with old versions of g++ that doesn't allow a structure
member to use the name list. Rename list member to group_list in
ast_group_info and change the few places it is used. (closes
issue #14790) Reported by: stuarth ........
* channels/h323/chan_h323.h, /, channels/h323/ast_h323.cxx,
channels/chan_h323.c: Merged revisions 189993 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r189993 |
jpeeler | 2009-04-22 14:23:49 -0500 (Wed, 22 Apr 2009) | 18 lines
Make chan_h323 respect packetization settings and fix small
reload issue. Previously, packetization settings were ignored and
now they are not. A new config option 'autoframing' has been
added to mirror the way chan_sip handles it. Turning on the
autoframing option (available both as a global option or per
peer) overrides the local settings with the remote packetization
settings. Testing was performed with varying packetization levels
with the following codecs: ulaw, alaw, gsm, and g729. Also, an
unrelated config reload issue has been fixed in the case of the
config file not changing. (closes issue #12415) Reported by: pj
Patches: 2009012200_h323packetization.diff.txt uploaded by
mvanbaak (license 7), modified by me ........
2009-04-22 18:01 +0000 [r189986] Russell Bryant <russell@digium.com>
* /, main/features.c: Merged revisions 189951 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r189951 |
russell | 2009-04-22 11:56:43 -0500 (Wed, 22 Apr 2009) | 2 lines
Fix call parking callback. Pipes -> Commas. ........
2009-04-22 16:04 +0000 [r189816-189914] Tilghman Lesher <tlesher@digium.com>
* channels/chan_unistim.c, /: Merged revisions 189911 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r189911 | tilghman | 2009-04-22 11:01:30 -0500 (Wed, 22 Apr 2009)
| 7 lines Do not continue to receive DTMF, when the channel is
hungup and about to be destroyed. (closes issue #14858) Reported
by: barryf Patches: 20090421__bug14858.diff.txt uploaded by
tilghman (license 14) Tested by: barryf ........
* /, configure, configure.ac: Merged revisions 189813 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r189813 | tilghman | 2009-04-22 01:33:08 -0500 (Wed, 22 Apr 2009)
| 3 lines Detect liblua on SuSE, and add libm for linking for
Fedora. (Reported via the -dev list, Subject: Compiling Asterisk
with LUA) ........
2009-04-21 20:45 +0000 [r189775] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 189771 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r189771 |
dvossel | 2009-04-21 15:28:37 -0500 (Tue, 21 Apr 2009) | 11 lines
Fixes segfault when switching UDP to TCP in sip.conf after
reload. If transport in sip.conf is switched from UDP to TCP,
Asterisk segfaults right after issuing a sip reload. The problem
is the socket type is changed to TCP but the fd may still be
present for UDP. Later, when the TCP session should be created or
set using an existing one, it isn't because the old file
descriptor is still present. Now every time transport is changed
during a sip.conf reload, the file descriptor is set to -1,
signifying it must be created or found. (closes issue #14727)
Reported by: pj Tested by: dvossel Review:
http://reviewboard.digium.com/r/229/ ........
2009-04-20 22:11 +0000 [r189540] Tilghman Lesher <tlesher@digium.com>
* main/stdtime/localtime.c, /: Merged revisions 189539 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r189539 | tilghman | 2009-04-20 17:10:25 -0500 (Mon, 20 Apr 2009)
| 3 lines Use nanosleep instead of poll. This is not just because
mmichelson suggested it, but also because Mac OS X puked on my
poll(). ........
2009-04-20 21:41 +0000 [r189536] Terry Wilson <twilson@digium.com>
* apps/app_dial.c, /: Merged revisions 189495,189516 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r189495 | twilson | 2009-04-20 16:24:34 -0500
(Mon, 20 Apr 2009) | 9 lines Merged revisions 189463 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r189463 | twilson | 2009-04-20 16:00:52 -0500 (Mon, 20
Apr 2009) | 2 lines Don't treat a NOANSWER like a CHANUNAVAIL
........ ................ r189516 | twilson | 2009-04-20 16:29:29
-0500 (Mon, 20 Apr 2009) | 9 lines Merged revisions 189465 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r189465 | twilson | 2009-04-20 16:10:27 -0500 (Mon, 20 Apr 2009)
| 2 lines Update CDR appropriately when AST_CAUSE_NO_ANSWER is
set ........ ................
2009-04-20 21:36 +0000 [r189533] Sean Bright <sean.bright@gmail.com>
* /, res/ael/ael.tab.c, res/ael/ael.y: Merged revisions 189464 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r189464 | seanbright | 2009-04-20 17:09:59 -0400
(Mon, 20 Apr 2009) | 20 lines Merged revisions 189462 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r189462 | seanbright | 2009-04-20 16:58:39 -0400 (Mon, 20 Apr
2009) | 13 lines Properly handle @s within hints in AEL. AEL was
not handling the case of a device hint containing an @ symbol,
which caused parking hints (e.g. hint(park:exten@context)) to
error out the parser. This patch makes AEL treat the @ the same
way it treats colon and ampersand now, meaning the characters are
included in verbatim. (closes issue #14941) Reported by: bpgoldsb
Patches: bug14941.patch uploaded by seanbright (license 71)
Tested by: bpgoldsb ........ ................
2009-04-20 17:11 +0000 [r189353] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 189350 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r189350 |
file | 2009-04-20 14:05:15 -0300 (Mon, 20 Apr 2009) | 10 lines
Fix a bug with non-UDP connections that caused dialogs to not get
freed. This issue crept up because of a reference count issue on
non-UDP based dialogs. The dialog reference count was increased
when transmitting a packet reliably but never decreased. This
caused the dialog structure to hang around despite being unlinked
from the dialogs container. (closes issue #14919) Reported by:
vrban ........
2009-04-20 14:07 +0000 [r189281] Mark Michelson <mmichelson@digium.com>
* main/channel.c, /: Merged revisions 189278 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r189278 | mmichelson | 2009-04-20 09:05:27 -0500 (Mon, 20 Apr
2009) | 18 lines Merged revisions 189277 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r189277 | mmichelson | 2009-04-20 09:04:41 -0500 (Mon, 20 Apr
2009) | 12 lines Move the check for chan->fdno == -1 to after the
zombie/hangup check. Many users were finding that their hung up
channels were staying up and causing 100% CPU usage. (issue
#14723) Reported by: seadweller Patches: 14723_1-4-tip.patch
uploaded by mmichelson (license 60) Tested by: falves11, bamby
........ ................
2009-04-18 01:42 +0000 [r189207-189208] David Vossel <dvossel@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 188647 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r188647 | dvossel | 2009-04-15 17:10:04 -0500
(Wed, 15 Apr 2009) | 18 lines Merged revisions 188646 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r188646 | dvossel | 2009-04-15 17:08:40 -0500 (Wed, 15 Apr 2009)
| 12 lines National prefix inserted even when caller ID not
available When the caller ID is restricted, the expected behavior
is for the caller id to be blank. In chan_dahdi, the national
prefix is placed onto the callers number even if its restricted
(empty) causing the caller id to be the national prefix rather
than blank. (closes issue #13207) Reported by: shawkris Patches:
national_prefix.diff uploaded by dvossel (license 671) Review:
http://reviewboard.digium.com/r/220/ ........ ................
* /, channels/chan_agent.c: Merged revisions 189204 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r189204 | dvossel | 2009-04-17 20:28:45 -0500
(Fri, 17 Apr 2009) | 18 lines Merged revisions 189203 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r189203 | dvossel | 2009-04-17 20:27:19 -0500 (Fri, 17 Apr 2009)
| 12 lines Fixed autologoff in agents.conf not working when agent
logs in via AgentLogin app An agent logs in by calling an
extension that calls the AgentLogin app. In agents.conf
ackcall=always is set, so when they get a call they have the
choice to either acknowledge it or ignore it. autologoff=10 is
set as well, so if the agent ignores the call over 10sec one may
assume that the agent should be logged out (and in this case
hungup on as well), but this was not happening. (closes issue
#14091) Reported by: evandro Patches: autologoff.diff uploaded by
dvossel (license 671) Review:
http://reviewboard.digium.com/r/225/ ........ ................
2009-04-17 21:56 +0000 [r189140] Richard Mudgett <rmudgett@digium.com>
* channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
revisions 189137 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r189137 | rmudgett | 2009-04-17 16:48:10 -0500 (Fri, 17 Apr 2009)
| 17 lines Merged revisions 188833,189134 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r188833 | rmudgett | 2009-04-16 16:37:58 -0500 (Thu, 16 Apr 2009)
| 4 lines Only disable mISDN DSP if Asterisk DSP is enabled.
Leave jitter setting alone. JIRA ABE-1835 ........ r189134 |
rmudgett | 2009-04-17 16:27:55 -0500 (Fri, 17 Apr 2009) | 4 lines
Modifed/added some debug messages. JIRA ABE-1835 ........
................
2009-04-17 20:21 +0000 [r189105] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 189097 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r189097 |
mmichelson | 2009-04-17 15:20:23 -0500 (Fri, 17 Apr 2009) | 13
lines Prevent a crash when SIP blonde transferring an unbridged
call. If one attempts to use the attended transfer button on a
SIP phone to transfer an unbridged call (such as a call to an
IVR) but hangs up while the target of the transfer is still
ringing, we need to not crash. The problem was that ast_hangup
was called from outside the channel thread. AST-211 ........
2009-04-17 19:47 +0000 [r189081] Sean Bright <sean.bright@gmail.com>
* main/asterisk.c, /: Merged revisions 189077 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r189077 |
seanbright | 2009-04-17 15:36:38 -0400 (Fri, 17 Apr 2009) | 1
line Fix copy/paste error with 'transmit silence' flag. ........
2009-04-17 17:31 +0000 [r189068] Matthew Nicholson <mnicholson@digium.com>
* main/pbx.c, /: Merged revisions 189010 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r189010 | mnicholson | 2009-04-17 10:44:18 -0500 (Fri, 17 Apr
2009) | 12 lines Merged revisions 189009 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r189009 | mnicholson | 2009-04-17 10:43:09 -0500 (Fri, 17 Apr
2009) | 5 lines Make Busy() application set the CDR disposition
to BUSY. (closes issue #14306) Reported by: cristiandimache
........ ................
2009-04-17 14:50 +0000 [r188941-188950] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 188947 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r188947 | file | 2009-04-17 11:44:56 -0300 (Fri, 17 Apr 2009) |
22 lines Merged revisions 188946 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r188946 | file | 2009-04-17 11:41:25 -0300 (Fri, 17 Apr 2009) |
15 lines Fix a bug where a value used to create the channel name
was bogus. This commit fixes the scenario where an incoming call
is authenticated using a peer entry. Previously the channel name
was created using either the username setting from the sip.conf
entry or the IP address that the call came from. Now the channel
name will be created using the peer name itself. This commit will
not change the way the channel name is generated for users or
friends. (closes issue #14256) Reported by: Nick_Lewis Patches:
chan_sip.c-chname.patch uploaded by Nick (license 657) Tested by:
Nick_Lewis, file ........ ................
* channels/chan_dahdi.c, /: Merged revisions 188938 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r188938 | file | 2009-04-17 11:26:53 -0300 (Fri,
17 Apr 2009) | 11 lines Merged revisions 188937 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r188937 | file | 2009-04-17 11:25:57 -0300 (Fri, 17 Apr 2009) | 4
lines Fix a situation where the DAHDI channel private structure
lock was not unlocked when it should have been. (issue AST-210)
........ ................
2009-04-16 22:05 +0000 [r188777-188839] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_sip.c: Merged revisions 188836 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r188836 | tilghman | 2009-04-16 16:57:37 -0500 (Thu, 16 Apr 2009)
| 14 lines Merged revisions 188835 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r188835 | tilghman | 2009-04-16 16:41:13 -0500 (Thu, 16 Apr 2009)
| 7 lines Only update realtime, if global option rtupdate !=
false (closes issue #14885) Reported by: deepesh Patches:
20090413__bug14885.diff.txt uploaded by tilghman (license 14)
Tested by: deepesh ........ ................
* apps/app_voicemail.c, /: Merged revisions 188774 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r188774 | tilghman | 2009-04-16 16:03:31 -0500
(Thu, 16 Apr 2009) | 11 lines Merged revisions 188773 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r188773 | tilghman | 2009-04-16 16:02:29 -0500 (Thu, 16 Apr 2009)
| 4 lines Umask should not be exported into global namespace.
(closes issue #14912) Reported by: jcapp ........
................
2009-04-15 20:20 +0000 [r188474-188598] Mark Michelson <mmichelson@digium.com>
* /, main/file.c: Merged revisions 188585 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r188585 | mmichelson | 2009-04-15 15:17:33 -0500 (Wed, 15 Apr
2009) | 13 lines Merged revisions 188582 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r188582 | mmichelson | 2009-04-15 15:04:20 -0500 (Wed, 15 Apr
2009) | 7 lines Update ast_readvideo_callback to match
ast_readaudio_callback. This fixes potential refcount errors that
may occur on ast_filestreams. AST-208 ........ ................
* apps/app_queue.c, /: Merged revisions 188470 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r188470 |
mmichelson | 2009-04-14 18:28:13 -0500 (Tue, 14 Apr 2009) | 3
lines Fix a couple of queue member reference leaks. ........
2009-04-14 17:46 +0000 [r188259-188416] Joshua Colp <jcolp@digium.com>
* main/rtp.c, /: Merged revisions 188413 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r188413 |
file | 2009-04-14 14:40:50 -0300 (Tue, 14 Apr 2009) | 5 lines Fix
an incorrect clock rate when sending T140 text. (closes issue
#14029) Reported by: epicac ........
* /, channels/chan_sip.c: Merged revisions 188247 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r188247 |
file | 2009-04-14 10:14:21 -0300 (Tue, 14 Apr 2009) | 7 lines Fix
a bug with the change I made yesterday to outbound proxy support.
Per discussion with oej on IRC we need the actual IP address, not
the outbound proxy IP address, in the sa field. Upon further
inspection this should make the behaviour of all other uses of
the outbound proxy in the code. ........
2009-04-14 05:47 +0000 [r188209-188213] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c, /: Merged revisions 188210 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r188210 |
tilghman | 2009-04-14 00:45:13 -0500 (Tue, 14 Apr 2009) | 2 lines
As suggested by Russell, warn users when their dialplan arguments
contain pipes, but not commas. ........
* /, utils/smsq.c: Merged revisions 188206 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r188206 |
tilghman | 2009-04-14 00:27:53 -0500 (Tue, 14 Apr 2009) | 6 lines
Application delimiter is ',', not '|'. (closes issue #14881)
Reported by: stegro Patches: smsq.patch uploaded by stegro
(license 752) ........
2009-04-13 19:33 +0000 [r188105] Mark Michelson <mmichelson@digium.com>
* res/res_musiconhold.c, /: Merged revisions 188102 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r188102 | mmichelson | 2009-04-13 14:31:48 -0500 (Mon, 13 Apr
2009) | 5 lines Fix another crash related to cached realtime
music on hold. This was another off-by-one problem caused by
moh_register. ........
2009-04-13 16:34 +0000 [r188070] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 188067 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r188067 |
file | 2009-04-13 13:28:06 -0300 (Mon, 13 Apr 2009) | 10 lines
Fix a bug where using an outbound proxy would cause the local
address to be 127.0.0.1. Copy the outbound proxy IP address into
the SIP dialog structure as the IP address we will be sending to.
This has to be done because the logic that determines what local
IP address to use in the SIP messages is not aware of an outbound
proxy being in place. It only knows what IP address we are
sending to. (closes issue #12006) Reported by: mnicholson
........
2009-04-13 14:20 +0000 [r188039] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c, /: Merged revisions 188032 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r188032 |
mmichelson | 2009-04-13 09:17:56 -0500 (Mon, 13 Apr 2009) | 6
lines Set all queue variables on both the caller and member
channels. This allows for the variables to be accessed if a
member macro is run. Thanks to Grigoriy Puzankin for bringing
this up on the -dev list. ........
2009-04-10 20:28 +0000 [r187916] Jeff Peeler <jpeeler@digium.com>
* channels/Makefile, /: Merged revisions 187906 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r187906 |
jpeeler | 2009-04-10 15:26:46 -0500 (Fri, 10 Apr 2009) | 12 lines
Fix module embedding for chan_h323. Include libchanh323.a in the
modules.link file so that all the symbols can be resolved at link
time. (closes issue #11966) Reported by: dome Patches:
issue_11966.patch uploaded by kpfleming (license 421) Tested by:
jpeeler ........
2009-04-10 17:31 +0000 [r187769] Tilghman Lesher <tlesher@digium.com>
* contrib/scripts/sip-friends.sql,
contrib/scripts/realtime_pgsql.sql, /: Merged revisions 187764
via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r187764 | tilghman | 2009-04-10 12:29:34 -0500
(Fri, 10 Apr 2009) | 9 lines Merged revisions 187763 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r187763 | tilghman | 2009-04-10 12:28:46 -0500 (Fri, 10
Apr 2009) | 2 lines Add lastms column to the contributed table
designs ........ ................
2009-04-10 16:54 +0000 [r187724] Kevin P. Fleming <kpfleming@digium.com>
* /, build_tools/embed_modules.xml: Merged revisions 187721 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r187721 | kpfleming | 2009-04-10 11:51:44 -0500 (Fri, 10
Apr 2009) | 5 lines clean up some patterns for files to remove
add embedding support for bridge and test modules ........
2009-04-10 16:05 +0000 [r187679] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_sip.c: Merged revisions 187674 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r187674 |
tilghman | 2009-04-10 10:59:40 -0500 (Fri, 10 Apr 2009) | 4 lines
Ensure pvt is not NULL before dereferencing it. (closes issue
#14784) Reported by: pj ........
2009-04-10 16:01 +0000 [r187677] Russell Bryant <russell@digium.com>
* tests/test_sched.c, tests/test_heap.c, /: Merged revisions 187675
via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r187675 | russell | 2009-04-10 11:00:29 -0500 (Fri, 10
Apr 2009) | 2 lines Disable test modules by default. ........
2009-04-10 03:57 +0000 [r187601] Tilghman Lesher <tlesher@digium.com>
* main/audiohook.c, main/bridging.c, main/channel.c, main/pbx.c,
main/manager.c, /, include/asterisk/linkedlists.h,
main/features.c, main/http.c, main/app.c,
include/asterisk/lock.h: Merged revisions 187599 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r187599 | tilghman | 2009-04-09 22:55:27 -0500 (Thu, 09 Apr 2009)
| 2 lines Modify headers and macros, according to Russell's
suggestions on the -dev list ........
2009-04-09 21:09 +0000 [r187564] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merge revision 187488 from trunk.
2009-04-09 19:29 +0000 [r187531-187546] David Vossel <dvossel@digium.com>
* main/audiohook.c, /: Merged revisions 186379 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r186379 |
dvossel | 2009-04-03 11:29:47 -0500 (Fri, 03 Apr 2009) | 6 lines
audio_audiohook_write_list() did not correctly update sample size
after ast_translate. audio_audiohook_write_list() did not take
into account that the sample size may change after translation
depending on if the original frame is is 8khz or 16khz. the
sample size is now updated after translating to reflect this
possibility. This caused the audio on the receiving end to sound
terrible. Thanks to jcolp and mmichelson for helping me work this
out. (issue AST-197) ........
* /, channels/chan_sip.c: Merged revisions 185846 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r185846 | dvossel | 2009-04-01 14:03:32 -0500 (Wed, 01 Apr 2009)
| 16 lines Merged revisions 185845 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r185845 | dvossel | 2009-04-01 14:02:00 -0500 (Wed, 01 Apr 2009)
| 10 lines Fixes issue with dropped calles due to re-Invite glare
and re-Invites never executing after a 491 Acknowledgement for
491 responses were never being processed because it didn't match
our pending invite's seqno. Since the ACK was never processed,
the 491 frame would continue to be retransmitted until eventually
the call was dropped due to max retries. Now during a pending
invite, if we receive another invite, we send an 491 and hold on
to that glare invite's seqno in the "glareinvite" variable for
that sip_pvt struct. When ACK's are received, we first check to
see if it is in response to our pending invite, if not we check
to see if it is in response to a glare invite. In this case, it
is in response to the glare invite and must be dealt with or the
call is dropped. I've changed the wait time for resending the
re-Invite after receving a 491 response to comply with RFC 3261.
Before this patch the scheduled re-Invite would only change a
flag indicating that the re-Invite should be sent out, now it
actually sends it out as well. (closes issue #12013) Reported by:
alx Review: http://reviewboard.digium.com/r/213/ ........
................
2009-04-09 19:15 +0000 [r187496] Mark Michelson <mmichelson@digium.com>
* res/res_musiconhold.c, /: Merged revisions 187421,187424 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r187421 | mmichelson | 2009-04-09 12:30:39 -0500 (Thu,
09 Apr 2009) | 21 lines Fix a crash in res_musiconhold when using
cached realtime moh. The moh_register function links an mohclass
and then immediately unrefs the class since the container now has
a reference. The problem with using realtime music on hold is
that the class is allocated, registered, and started in one fell
swoop. The refcounting logic resulted in the count being off by
one. The same problem did not happen when using a static config
because the allocation and registration of an mohclass is a
separate operation from starting moh. This also did not affect
non-cached realtime moh because the classes are not registered at
all. I also have modified res_musiconhold to use the _t_ variants
of the ao2_ functions so that more info can be gleaned when
attempting to trace the refcounts. I found this to be incredibly
helpful for debugging this issue and there's no good reason to
remove it. (closes issue #14661) Reported by: sum ........
r187424 | mmichelson | 2009-04-09 12:34:39 -0500 (Thu, 09 Apr
2009) | 3 lines Use safe macro practices even though they really
aren't necessary. ........
2009-04-09 18:55 +0000 [r187051-187487] Tilghman Lesher <tlesher@digium.com>
* main/manager.c, /, include/asterisk/linkedlists.h,
include/asterisk/lock.h: Merged revisions 187483 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r187483 | tilghman | 2009-04-09 13:40:01 -0500
(Thu, 09 Apr 2009) | 15 lines Merged revisions 187428 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r187428 | tilghman | 2009-04-09 13:08:20 -0500 (Thu, 09 Apr 2009)
| 8 lines Race condition between ast_cli_command() and 'module
unload' could cause a deadlock. Add lock timeouts to avoid this
potential deadlock. (closes issue #14705) Reported by: jamessan
Patches: 20090320__bug14705.diff.txt uploaded by tilghman
(license 14) Tested by: jamessan ........ ................
* /, channels/chan_sip.c: Merged revisions 187381 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r187381 |
tilghman | 2009-04-09 12:20:49 -0500 (Thu, 09 Apr 2009) | 4 lines
Allow '/' in username portion of register; this is a regression.
(closes issue #14668) Reported by: Netview ........
* /, channels/chan_sip.c, apps/app_sendtext.c: Merged revisions
187363 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r187363 | tilghman | 2009-04-09 11:39:43 -0500 (Thu, 09 Apr 2009)
| 10 lines Merged revisions 187362 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r187362 | tilghman | 2009-04-09 11:38:37 -0500 (Thu, 09 Apr 2009)
| 3 lines Permit zero-length text messages in SIP. (Related to an
issue posted to the -users list, subject "AEL2, BASE64_DECODE and
hexadecimal") ........ ................
* main/asterisk.c, agi/Makefile, build_tools/cflags.xml,
utils/Makefile, include/asterisk.h, /, main/Makefile,
main/file.c, main/astfd.c (added): Merged revisions 187302 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r187302 | tilghman | 2009-04-08 23:59:05 -0500
(Wed, 08 Apr 2009) | 14 lines Merged revisions 187300-187301 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r187300 | tilghman | 2009-04-08 23:31:38 -0500 (Wed, 08 Apr 2009)
| 3 lines Add debugging mode for diagnosing file descriptor
leaks. (Related to issue #14625) ........ r187301 | tilghman |
2009-04-08 23:32:40 -0500 (Wed, 08 Apr 2009) | 2 lines Oops,
missed this file in the last commit. ........ ................
* /, funcs/func_odbc.c: Merged revisions 187050 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r187050 |
tilghman | 2009-04-08 12:08:43 -0500 (Wed, 08 Apr 2009) | 7 lines
If the first column is empty, output a delimiter anyway. (closes
issue #14848) Reported by: john8675309 Patches:
20090408__bug14848.diff.txt uploaded by tilghman (license 14)
Tested by: john8675309 ........
2009-04-08 16:54 +0000 [r186988-187049] Mark Michelson <mmichelson@digium.com>
* res/res_musiconhold.c, /: Merged revisions 187046 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r187046 | mmichelson | 2009-04-08 11:52:20 -0500
(Wed, 08 Apr 2009) | 16 lines Merged revisions 187045 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r187045 | mmichelson | 2009-04-08 11:52:03 -0500 (Wed, 08 Apr
2009) | 10 lines Fix a small logical error when loading moh
classes. We were unconditionally incrementing the number of
mohclasses registered. However, we should actually only increment
if the call to moh_register was successful. While this probably
has never caused problems, I noticed it and decided to fix it
anyway. ........ ................
* main/channel.c, /: Merged revisions 186985 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r186985 | mmichelson | 2009-04-08 10:27:41 -0500 (Wed, 08 Apr
2009) | 30 lines Merged revisions 186984 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r186984 | mmichelson | 2009-04-08 10:26:46 -0500 (Wed, 08 Apr
2009) | 24 lines Make a couple of changes with regards to a new
message printed in ast_read(). "ast_read() called with no
recorded file descriptor" is a new message added after a bug was
discovered. Unfortunately, it seems there are a bunch of places
that potentially make such calls to ast_read() and trigger this
error message to be displayed. This commit does two things to
help to make this message appear less. First, the message has
been downgraded to a debug level message if dev mode is not
enabled. The message means a lot more to developers than it does
to end users, and so developers should take an effort to be sure
to call ast_read only when a channel is ready to be read from.
However, since this doesn't actually cause an error in operation
and is not something a user can easily fix, we should not spam
their console with these messages. Second, the message has been
moved to after the check for any pending masquerades. ast_read()
being called with no recorded file descriptor should not
interfere with a masquerade taking place. This could be seen as a
simple way of resolving issue #14723. However, I still want to
try to clear out the existing ways of triggering this message,
since I feel that would be a better resolution for the issue.
........ ................
2009-04-08 12:39 +0000 [r186929] Russell Bryant <russell@digium.com>
* /, channels/chan_sip.c: Merged revisions 186928 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r186928 |
russell | 2009-04-08 07:35:57 -0500 (Wed, 08 Apr 2009) | 13 lines
Update some comments and resolve potential memory corruption in
chan_sip. While browsing chan_sip the other day, I noticed this
dangerous code in dialog_needdestroy(). This function is an
ao2_callback. It is absolutely _not_ okay to unlock the container
from within this function. It's also not clear why it was useful.
Given that it could cause memory corruption, I have removed it.
There was also a TODO comment left describing a potential
implementation of an improvement to the needdestroy handling. I'm
not convinced that what was described is the best choice here, so
I have briefly described the way that this function is used today
that could be improved. ........
2009-04-08 05:08 +0000 [r186901] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_sip.c: Merged revisions 186899 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r186899 |
tilghman | 2009-04-08 00:06:22 -0500 (Wed, 08 Apr 2009) | 2 lines
Add lastms to the require API call. ........
2009-04-08 00:10 +0000 [r186836-186845] Mark Michelson <mmichelson@digium.com>
* formats/format_wav_gsm.c, /, formats/format_wav.c: Merged
revisions 186842 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r186842 | mmichelson | 2009-04-07 19:09:28 -0500 (Tue, 07 Apr
2009) | 14 lines Merged revisions 186841 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r186841 | mmichelson | 2009-04-07 19:09:04 -0500 (Tue, 07 Apr
2009) | 8 lines Fix a few typos of the word "frequency." (closes
issue #14842) Reported by: jvandal Patches: frequency-typo.diff
uploaded by jvandal (license 413) ........ ................
* /, channels/chan_sip.c: Merged revisions 186837 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r186837 |
mmichelson | 2009-04-07 19:01:49 -0500 (Tue, 07 Apr 2009) | 7
lines Fix bad merge from fix for issue 13867. (closes issue
#14686) Reported by: davidw ........
* main/channel.c, /: Merged revisions 186833 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r186833 | mmichelson | 2009-04-07 18:50:56 -0500 (Tue, 07 Apr
2009) | 15 lines Merged revisions 186832 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r186832 | mmichelson | 2009-04-07 18:49:49 -0500 (Tue, 07 Apr
2009) | 8 lines Set the AST_FEATURE_WARNING_ACTIVE flag when a
p2p bridge returns AST_BRIDGE_RETRY. Without this flag set,
warning sounds will not be properly played to either party of the
bridge. (closes issue #14845) Reported by: adomjan ........
................
2009-04-07 22:33 +0000 [r186807] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_macro.c: Merged revisions 186799 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r186799 | tilghman | 2009-04-07 17:23:46 -0500 (Tue, 07 Apr 2009)
| 10 lines Merged revisions 186775 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r186775 | tilghman | 2009-04-07 17:16:50 -0500 (Tue, 07 Apr 2009)
| 3 lines Fix Macro documentation to match current (and intended)
behavior. (See -dev mailing list) ........ ................
2009-04-07 20:59 +0000 [r186723] Mark Michelson <mmichelson@digium.com>
* main/manager.c, /: Merged revisions 186720 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r186720 | mmichelson | 2009-04-07 15:46:18 -0500 (Tue, 07 Apr
2009) | 12 lines Merged revisions 186719 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r186719 | mmichelson | 2009-04-07 15:43:49 -0500 (Tue, 07 Apr
2009) | 6 lines Ensure that \r\n is printed after the ActionID in
an OriginateResponse. (closes issue #14847) Reported by: kobaz
........ ................
2009-04-03 20:21 +0000 [r186469] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 186461 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r186461 | kpfleming | 2009-04-03 15:20:01 -0500
(Fri, 03 Apr 2009) | 11 lines Merged revisions 186458 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r186458 | kpfleming | 2009-04-03 15:19:20 -0500 (Fri, 03 Apr
2009) | 5 lines Fix a bug where DAHDI/Zaptel channels would not
properly switch formats when requested Don't offer
AST_FORMAT_SLINEAR on DAHDI/Zaptel channels... while it could
provide a slight performance benefit, the translation core in
Asterisk has some flaws when a channel driver offers multiple raw
formats. this fix is much simpler than fixing the translation
core to solve that issue (although that will be done later).
........ ................
2009-04-03 20:05 +0000 [r186449] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c, /, configs/voicemail.conf.sample: Merged
revisions 186444,186447 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r186444 | tilghman | 2009-04-03 14:30:34 -0500 (Fri, 03 Apr 2009)
| 14 lines Merged revisions 186415 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r186415 | tilghman | 2009-04-03 14:06:58 -0500 (Fri, 03 Apr 2009)
| 7 lines Distinguish in a sent email between simple sends and
forwards. (closes issue #11678) Reported by: jamessan Patches:
20090330__bug11678.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman, lmadsen ........ ................ r186447 |
tilghman | 2009-04-03 14:59:55 -0500 (Fri, 03 Apr 2009) | 9 lines
Merged revisions 186445 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r186445 | tilghman | 2009-04-03 14:56:48 -0500 (Fri, 03 Apr 2009)
| 2 lines Found a conflict in the last commit, due to multiple
targets ........ ................
2009-04-03 15:56 +0000 [r186324] Joshua Colp <jcolp@digium.com>
* include/asterisk/crypto.h, /: Merged revisions 186321 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r186321 | file | 2009-04-03 12:52:50 -0300 (Fri,
03 Apr 2009) | 12 lines Merged revisions 186320 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r186320 | file | 2009-04-03 12:48:56 -0300 (Fri, 03 Apr 2009) | 5
lines Fix a problem with the crypto variable definitions not
actually being defined properly. (closes issue #14804) Reported
by: jvandal ........ ................
2009-04-03 15:19 +0000 [r186302] Tilghman Lesher <tlesher@digium.com>
* main/stdtime/localtime.c, /: Merged revisions 186297 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r186297 | tilghman | 2009-04-03 10:18:28 -0500 (Fri, 03 Apr 2009)
| 4 lines Compatibility fix for glibc 2.4 (Closes issue #14820)
Reported by: phsultan ........
2009-04-03 14:34 +0000 [r186289] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c, /: Merged revisions 186286 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r186286 | mmichelson | 2009-04-03 09:32:05 -0500 (Fri, 03 Apr
2009) | 20 lines Fix the ability to retrieve voicemail messages
from IMAP. A recent change made interactive vm_states no longer
get added to the list of vm_states and instead get stored in
thread-local storage. In trunk and all the 1.6.X branches, the
problem is that when we search for messages in a voicemail box,
we would attempt to update the appropriate vm_state struct by
directly searching in the list of vm_states instead of using the
get_vm_state_by_imap_user function. This meant we could not find
the interactive vm_state that we wanted. (closes issue #14685)
Reported by: BlargMaN Patches: 14685.patch uploaded by mmichelson
(license 60) Tested by: BlargMaN, qualleyiv, mmichelson ........
2009-04-03 02:11 +0000 [r186233] Russell Bryant <russell@digium.com>
* cdr/cdr_radius.c, /: Merged revisions 186230 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r186230 | russell | 2009-04-02 21:03:48 -0500 (Thu, 02 Apr 2009)
| 29 lines Merged revisions 186229 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r186229 | russell | 2009-04-02 20:57:44 -0500 (Thu, 02 Apr 2009)
| 21 lines Fix a memory leak in cdr_radius. I came across this
while doing some testing of my ast_channel_ao2 branch. After
running a test overnight that generated over 5 million calls,
Asterisk had taken up about 1 GB of my system memory. So, I
re-ran the test with MALLOC_DEBUG turned on. However, it showed
no leaks in Asterisk during the test, even though Asterisk was
still consuming it somehow. Instead, I turned to valgrind, which
when run with --leak-check=full, told me exactly where the leak
came from, which was from allocations inside the radiusclient-ng
library. This explains why MALLOC_DEBUG did not report it. After
a bit of analysis, I found that we were leaking a little bit of
memory every time a CDR record was passed to cdr_radius. I don't
actually have a radius server set up to receive CDR records.
However, I always have my development systems compile and install
all modules. In addition to making sure there are not build
errors across modules, always loading modules helps find bugs
like this, too, so it is strongly recommend for all developers.
........ ................
2009-04-02 22:00 +0000 [r186178] Mark Michelson <mmichelson@digium.com>
* configs/features.conf.sample, /: Merged revisions 186175 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r186175 | mmichelson | 2009-04-02 16:56:21 -0500
(Thu, 02 Apr 2009) | 11 lines Merged revisions 186174 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r186174 | mmichelson | 2009-04-02 16:55:34 -0500 (Thu, 02 Apr
2009) | 5 lines Fix instructions in one-step parking comment to
make more sense. Changed a capital K to a lowercase k. ........
................
2009-04-02 17:28 +0000 [r186111] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 186101 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r186101 | kpfleming | 2009-04-02 12:26:07 -0500
(Thu, 02 Apr 2009) | 9 lines Merged revisions 186081 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r186081 | kpfleming | 2009-04-02 12:21:29 -0500 (Thu, 02
Apr 2009) | 3 lines ensure that the buffer passed to
DAHDI_SET_BUFINFO is fully initialized ........ ................
2009-04-02 17:14 +0000 [r186022-186063] Tilghman Lesher <tlesher@digium.com>
* configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions
186060 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r186060 | tilghman | 2009-04-02 12:10:28 -0500 (Thu, 02 Apr 2009)
| 16 lines Merged revisions 186059 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
................ r186059 | tilghman | 2009-04-02 12:09:13 -0500
(Thu, 02 Apr 2009) | 9 lines Merged revisions 186056 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r186056 | tilghman | 2009-04-02 12:02:18 -0500 (Thu, 02
Apr 2009) | 2 lines Fix for AST-2009-003 ........
................ ................
* main/strings.c, /: Merged revisions 186021 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r186021 |
tilghman | 2009-04-02 10:14:22 -0500 (Thu, 02 Apr 2009) | 7 lines
Missed a common case for needing to extend the buffer. (closes
issue #14716) Reported by: sum Patches:
20090402__bug14716.diff.txt uploaded by tilghman (license 14)
Tested by: sum ........
2009-04-02 13:54 +0000 [r185957] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 185953 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r185953 | kpfleming | 2009-04-02 08:51:44 -0500
(Thu, 02 Apr 2009) | 11 lines Merged revisions 185952 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r185952 | kpfleming | 2009-04-02 08:43:43 -0500 (Thu, 02 Apr
2009) | 5 lines the DAHDI_GETCONF, DAHDI_SETCONF and
DAHDI_GET_PARAMS ioctls were recently corrected to show that they
do, in fact, read data from userspace as part of their work. due
to this fix, valgrind now reports a number of cases where
chan_dahdi passed an uninitialized (or partially) buffer to these
ioctls, which could lead to unexpected behavior. this patch
corrects chan_dahdi to ensure that buffers passed to these ioctls
are always fully initialized. ........ ................
2009-04-01 22:44 +0000 [r185947] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/pbx.h, include/asterisk/strings.h,
main/taskprocessor.c, res/res_odbc.c,
include/asterisk/res_odbc.h, include/asterisk.h, main/strings.c,
main/manager.c, /, main/tdd.c, include/asterisk/astobj2.h,
main/ast_expr2f.c: Merged revisions 185912 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r185912 |
tilghman | 2009-04-01 15:13:28 -0500 (Wed, 01 Apr 2009) | 4 lines
Merge changes from str_substitution that are unrelated to that
branch. Included is a small bugfix to an ast_str helper, but most
of these changes are simply doxygen fixes. ........
2009-04-01 13:51 +0000 [r185775] Russell Bryant <russell@digium.com>
* main/channel.c, /: Merged revisions 185772 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r185772 | russell | 2009-04-01 08:48:26 -0500 (Wed, 01 Apr 2009)
| 14 lines Merged revisions 185771 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r185771 | russell | 2009-04-01 08:47:30 -0500 (Wed, 01 Apr 2009)
| 6 lines Fix a case where DTMF could bypass audiohooks. This
change fixes a situation where an audiohook that wants DTMF would
not actually get it. This is in the code path where we end DTMF
digit length emulation while handling a NULL frame. ........
................
2009-03-31 22:38 +0000 [r185667] Kevin P. Fleming <kpfleming@digium.com>
* utils, /: Merged revisions 185664 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r185664 |
kpfleming | 2009-03-31 17:35:07 -0500 (Tue, 31 Mar 2009) | 1 line
ignore copied (generated) file ........
2009-03-31 22:13 +0000 [r185472-185605] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c, /: Merged revisions 185604 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r185604 |
mmichelson | 2009-03-31 17:12:52 -0500 (Tue, 31 Mar 2009) | 3
lines Fix trunk's compilation. ........
* apps/app_queue.c, /: Merged revisions 185600 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r185600 | mmichelson | 2009-03-31 17:02:48 -0500 (Tue, 31 Mar
2009) | 12 lines Merged revisions 185599 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r185599 | mmichelson | 2009-03-31 17:00:01 -0500 (Tue, 31 Mar
2009) | 6 lines Fix crash that would occur if an empty member was
specified in queues.conf. (closes issue #14796) Reported by: pida
........ ................
* apps/app_voicemail.c, /: Merged revisions 185469 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r185469 | mmichelson | 2009-03-31 14:46:18 -0500
(Tue, 31 Mar 2009) | 14 lines Merged revisions 185468 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r185468 | mmichelson | 2009-03-31 14:45:30 -0500 (Tue, 31 Mar
2009) | 8 lines Fix Russian voicemail intro to say the word
"messages" properly. (closes issue #14736) Reported by: chappell
Patches: voicemail_no_messages.diff uploaded by chappell (license
8) ........ ................
2009-03-31 17:51 +0000 [r185428] David Brooks <dbrooks@digium.com>
* channels/chan_gtalk.c, /: Merged revisions 185363 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r185363 | dbrooks | 2009-03-31 11:46:57 -0500
(Tue, 31 Mar 2009) | 44 lines Merged revisions 185362 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r185362 | dbrooks | 2009-03-31 11:37:12 -0500 (Tue, 31 Mar 2009)
| 35 lines Fix incorrect parsing in chan_gtalk when xmpp contains
extra whitespaces To drill into the xmpp to find the capabilities
between channels, chan_gtalk calls iks_child() and iks_next().
iks_child() and iks_next() are functions in the iksemel xml
parsing library that traverse xml nodes. The bug here is that
both iks_child() and iks_next() will return the next iks_struct
node *regardless* of type. chan_gtalk expects the next node to be
of type IKS_TAG, which in most cases, it is, but in this case (a
call being made from the Empathy IM client), there exists
iks_struct nodes which are not IKS_TAG data (they are extraneous
whitespaces), and chan_gtalk doesn't handle that case, so
capabilities don't match, and a call cannot be made.
iks_first_tag() and iks_next_tag(), on the other hand, will not
return the very next iks_struct, but will check to see if the
next iks_struct is of type IKS_TAG. If it isn't, it will be
skipped, and the next struct of type IKS_TAG it finds will be
returned. This assures that chan_gtalk will find the iks_struct
it is looking for. This fix simply changes all calls to
iks_child() and iks_next() to become calls to iks_first_tag() and
iks_next_tag(), which resolves the capability matching. The
following is a payload listing from Empathy, which, due to the
extraneous whitespace, will not be parsed correctly by iksemel:
<iq from='dbrooksjab@235-22-24-10/Telepathy'
to='astjab@235-22-24-10/asterisk' type='set' id='542757715704'>
<session xmlns='http://www.google.com/session'
initiator='dbrooksjab@235-22-24-10/Telepathy' type='initiate'
id='1837267342'> <description
xmlns='http://www.google.com/session/phone'> <payload-type
clockrate='16000' name='speex' id='96'/> <payload-type
clockrate='8000' name='PCMA' id='8'/> <payload-type
clockrate='8000' name='PCMU' id='0'/> <payload-type
clockrate='90000' name='MPA' id='97'/> <payload-type
clockrate='16000' name='SIREN' id='98'/> <payload-type
clockrate='8000' name='telephone-event' id='99'/> </description>
</session> </iq> Review: http://reviewboard.digium.com/r/181/
........ ................
2009-03-31 14:59 +0000 [r185264] Russell Bryant <russell@digium.com>
* apps/app_queue.c, /: Merged revisions 185261 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r185261 |
russell | 2009-03-31 09:53:45 -0500 (Tue, 31 Mar 2009) | 5 lines
Don't free() an astobj2 object. (closes issue #14672) Reported
by: makoto ........
2009-03-31 14:11 +0000 [r185200] Joshua Colp <jcolp@digium.com>
* main/audiohook.c, /: Merged revisions 185197 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r185197 | file | 2009-03-31 11:07:36 -0300 (Tue, 31 Mar 2009) |
15 lines Merged revisions 185196 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r185196 | file | 2009-03-31 11:06:39 -0300 (Tue, 31 Mar 2009) | 8
lines Fix crash when moving audiohooks between channels. Handle
the scenario where we are called to move audiohooks between
channels and the source channel does not actually have any on it.
(closes issue #14734) Reported by: corruptor ........
................
2009-03-30 20:52 +0000 [r185128-185129] Richard Mudgett <rmudgett@digium.com>
* channels/misdn_config.c, /, configs/misdn.conf.sample: Merged
revisions 185123 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r185123 | rmudgett | 2009-03-30 15:42:14 -0500 (Mon, 30 Mar 2009)
| 9 lines Merged revisions 185121 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r185121 | rmudgett | 2009-03-30 15:40:11 -0500 (Mon, 30 Mar 2009)
| 1 line Update the channel allocation method documentation.
........ ................
* channels/misdn/isdn_lib.c, /: Merged revisions 185122 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r185122 | rmudgett | 2009-03-30 15:41:24 -0500
(Mon, 30 Mar 2009) | 26 lines Merged revisions 185120 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r185120 | rmudgett | 2009-03-30 15:38:11 -0500 (Mon, 30 Mar 2009)
| 19 lines Make chan_misdn BRI TE side normally defer channel
selection to the NT side. Channel allocation collisions are not
handled by chan_misdn very well. This patch simply avoids the
problem for BRI only. For PRI, allocation collisions are still
possible but less likely since there are simply more channels
available and each end could use a different allocation strategy.
misdn.conf options available: te_choose_channel - Use to force
the TE side to allocate channels. method - Specify the channel
allocation strategy. (closes issue #13488) Reported by:
Christian_Pinedo Patches: isdn_lib.patch.txt uploaded by crich
Tested by: crich, siepkes, festr ........ ................
2009-03-30 16:52 +0000 [r185089] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c, /: Merged revisions 185072 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r185072 | mmichelson | 2009-03-30 11:26:48 -0500 (Mon, 30 Mar
2009) | 45 lines Merged revisions 185031 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r185031 | mmichelson | 2009-03-30 11:17:35 -0500 (Mon, 30 Mar
2009) | 39 lines Fix queue weight behavior so that calls in
low-weight queues are not inappropriately blocked. (This is
copied and pasted from the review request I made for this patch)
Asterisk has some odd behavior when queue weights are used. The
current logic used when potentially calling a queue member is: If
the member we are going to call is part of another queue and
_that other queue has any callers in it_ and has a higher weight
than the queue we are calling from, then don't try to contact
that member. The issue here is what I have marked with
underscores. If the higher-weighted queue has any callers in it
at all, then the queue member will be unreachable from the
lower-weighted queue. This has the potential to be really really
bad if using a queue strategy, such as leastrecent or
fewestcalls, with the potential to call the same member
repeatedly. The fix proposed by garychen on issue 13220 is very
simple and, as far as I can see, works well for this situation.
With this set of changes, the logic used becomes: If the member
we are going to call is part of another queue, the other queue
has a higher weight than the queue we are calling from, and the
higher weight queue has at least as many callers as available
members, then do not try to contact the queue member. If the
higher weighted queue has fewer callers than available members,
then there is no reason to deny the call to this member since the
other queue can afford to spare a member. Since the fix involved
writing a generic function for determining the number of
available members in the queue, I also modified the is_our_turn
function to make use of the new num_available_members function to
determine if it is our turn to try calling a member. There is one
small behavior change. Before writing this patch, if you had
autofill disabled, then if you were the head caller in a queue,
you would automatically be told that it was your turn to try
calling a member. This did not take into account whether there
were actually any queue members available to take the call. Now
we actually make sure there is at least one member available to
take the call if autofill is disabled. (closes issue #13220)
Reported by: garychen Review:
http://reviewboard.digium.com/r/202/ ........ ................
2009-03-30 14:43 +0000 [r184951] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 184948 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r184948 | file | 2009-03-30 11:37:47 -0300 (Mon, 30 Mar 2009) |
21 lines Merged revisions 184947 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r184947 | file | 2009-03-30 11:35:47 -0300 (Mon, 30 Mar 2009) |
14 lines Improve our handling of T38 in the initial INVITE from a
device. We now answer with matching media streams to what is
requested. If an INVITE is received with both a T38 and RTP media
stream this means we answer with both. For any outgoing calls
created as a result of this inbound one no T38 is requested in
the initial INVITE. Instead if we start receiving udptl packets
we trigger a reinvite on the outbound side. (closes issue #12437)
Reported by: marsosa Tested by: pinga-fogo, okrief, file, afu
Review: http://reviewboard.digium.com/r/208/ ........
................
2009-03-30 13:57 +0000 [r184913] Russell Bryant <russell@digium.com>
* channels/h323/Makefile.in, /: Merged revisions 184910 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r184910 | russell | 2009-03-30 08:55:44 -0500 (Mon, 30
Mar 2009) | 4 lines Fix build error when chan_h323 is not being
built. (reported by cai1982 in #asterisk-dev) ........
2009-03-29 05:56 +0000 [r184839-184846] Russell Bryant <russell@digium.com>
* apps/app_followme.c, /: Merged revisions 184843 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r184843 | russell | 2009-03-29 00:52:20 -0500 (Sun, 29 Mar 2009)
| 13 lines Merged revisions 184842 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r184842 | russell | 2009-03-29 00:51:55 -0500 (Sun, 29 Mar 2009)
| 5 lines Ensure targs variable is fully initialized. (closes
issue #14758) Reported by: tim_ringenbach ........
................
* channels/Makefile, /: Merged revisions 184838 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r184838 |
russell | 2009-03-29 00:32:04 -0500 (Sun, 29 Mar 2009) | 8 lines
Simplify chan_h323 build to not require a second run of "make".
(closes issue #14715) Reported by: jthurman Patches:
h323-makefile-1.6.2.0-beta1.patch uploaded by jthurman (license
614) Tested by: tzafrir, russell ........
2009-03-27 19:21 +0000 [r184779] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_iax2.c, main/timing.c, main/channel.c, /,
bridges/bridge_softmix.c, include/asterisk/timing.h,
include/asterisk/channel.h: Merged revisions 184762 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r184762 | kpfleming | 2009-03-27 14:10:32 -0500 (Fri, 27 Mar
2009) | 12 lines Improve timing interface to remember which
provider provided a timer The ability to load/unload timing
interfaces is nice, but it means that when a timer is allocated,
it may come from provider A, but later provider B becomes the
'preferred' provider. If this happens, all timer API calls on the
timer that was provided by provider A will actually be handed to
provider B, which will say WTF and return an error. This patch
changes the timer API to include a pointer to the provider of the
timer handle so that future operations on the timer will be
forwarded to the proper provider. (closes issue #14697) Reported
by: moy Review: http://reviewboard.digium.com/r/211/ ........
2009-03-27 18:12 +0000 [r184707-184729] Russell Bryant <russell@digium.com>
* main/manager.c, /, apps/app_minivm.c: Merged revisions 184726 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r184726 | russell | 2009-03-27 13:04:43 -0500 (Fri, 27
Mar 2009) | 2 lines Use ast_random() instead of rand() to ensure
we use the best RNG available. ........
* apps/app_queue.c, apps/app_voicemail.c, main/cli.c,
include/asterisk/app.h, /, apps/app_dumpchan.c, main/app.c:
Merged revisions 184693 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r184693 |
russell | 2009-03-27 11:21:10 -0500 (Fri, 27 Mar 2009) | 2 lines
Change global_app_buf to ast_str_thread_global_buf. ........
2009-03-27 15:58 +0000 [r184650-184678] Joshua Colp <jcolp@digium.com>
* /, bridges/bridge_softmix.c: Merged revisions 184677 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r184677 | file | 2009-03-27 12:57:28 -0300 (Fri, 27 Mar 2009) | 7
lines Fix a potential timer leak in bridge_softmix. It is
possible for a bridge to be created without actually being used.
In that scenario a timing file descriptor would be opened and not
closed. To fix this the timing file descriptor is now closed in
the destroy callback, not the thread function. ........
* /, res/res_agi.c: Merged revisions 184673 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r184673 |
file | 2009-03-27 12:46:46 -0300 (Fri, 27 Mar 2009) | 7 lines Fix
speech structure leak in the AGI speech recognition integration.
The AGI dialplan applications did not destroy the speech
structure automatically if it was not destroyed by the running
AGI script. They will now do this. (issue LUMENVOX-15) ........
* /, bridges/bridge_softmix.c: Merged revisions 184639 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r184639 | file | 2009-03-27 11:18:40 -0300 (Fri, 27 Mar 2009) | 2
lines Remove a cast that is not needed. ........
2009-03-27 14:09 +0000 [r184632] Russell Bryant <russell@digium.com>
* main/asterisk.c, include/asterisk/utils.h, main/pbx.c, /,
res/ais/evt.c, main/event.c, pbx/pbx_dundi.c: Merged revisions
184630 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r184630 |
russell | 2009-03-27 09:00:18 -0500 (Fri, 27 Mar 2009) | 2 lines
Change g_eid to ast_eid_default. ........
2009-03-27 13:59 +0000 [r184612-184629] Joshua Colp <jcolp@digium.com>
* /, bridges/bridge_softmix.c: Merged revisions 184628 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r184628 | file | 2009-03-27 10:57:29 -0300 (Fri, 27 Mar 2009) | 6
lines Fix a potential race condition when creating a software
based mixing bridge. It was possible for no timer to become
available between creating the bridge and starting it. We now
open a timer when creating it and keep it open until the bridge
is destroyed. ........
* /, channels/chan_sip.c: Merged revisions 184566 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r184566 | file | 2009-03-27 10:15:26 -0300 (Fri, 27 Mar 2009) |
16 lines Merged revisions 184565 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r184565 | file | 2009-03-27 10:06:45 -0300 (Fri, 27 Mar 2009) | 9
lines Fix an issue where nat=yes would not always take effect for
the RTP session on outgoing calls. If calls were placed using an
IP address or hostname the global nat setting was copied over but
was not set on the RTP session itself. This caused the RTP stack
to not perform symmetric RTP actions. (closes issue #14546)
Reported by: acunningham ........ ................
2009-03-27 02:35 +0000 [r184514-184552] Russell Bryant <russell@digium.com>
* /, include/asterisk/lock.h: Merged revisions 184531 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r184531 | russell | 2009-03-26 21:20:23 -0500 (Thu, 26 Mar 2009)
| 20 lines Fix some issues with rwlock corruption that caused
deadlock like symptoms. When dvossel and I were doing some load
testing last week, we noticed that we could make Asterisk trunk
lock up instantly when we started generating a bunch of calls.
The backtraces of locked threads were bizarre, and many were
stuck on an _unlock_ of an rwlock. The changes are: 1) Fix a
number of places where a backtrace would be loaded into an
invalid index of the backtrace array. It's an off by one error,
which ends up writing over the rwlock itself. 2) Ensure that in
the array of held locks, we NULL out an index once it is not
being used so that it's not confusing when analyzing its
contents. 3) Remove a bunch of logging referring to an rwlock
operating being done with "deep reentrancy". It is normal for
_many_ threads to hold a read lock on an rwlock. ........
* /, main/file.c: Merged revisions 184515 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r184515 |
russell | 2009-03-26 20:40:28 -0500 (Thu, 26 Mar 2009) | 2 lines
Don't act surprised if we get a -1 indication. ........
* include/asterisk/heap.h, /, main/heap.c: Merged revisions 184512
via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r184512 | russell | 2009-03-26 20:35:56 -0500 (Thu, 26
Mar 2009) | 2 lines Pass more useful information through to lock
tracking when DEBUG_THREADS is on. ........
2009-03-26 22:19 +0000 [r184454] Kevin P. Fleming <kpfleming@digium.com>
* sounds/Makefile, /: Merged revisions 184448 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r184448 | kpfleming | 2009-03-26 17:18:14 -0500 (Thu, 26 Mar
2009) | 9 lines Merged revisions 184447 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r184447 | kpfleming | 2009-03-26 17:17:32 -0500 (Thu, 26 Mar
2009) | 3 lines use new, improved 8kHz prompts ........
................
2009-03-25 22:15 +0000 [r184343-184346] Russell Bryant <russell@digium.com>
* /, main/event.c: Merged revisions 184344 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r184344 |
russell | 2009-03-25 17:11:35 -0500 (Wed, 25 Mar 2009) | 2 lines
Remove unneeded AST_LIST_ENTRY() and comment on the purpose of
ast_event_ref. ........
* include/asterisk/_private.h, channels/chan_iax2.c,
channels/chan_dahdi.c, include/asterisk/event.h,
apps/app_minivm.c, res/ais/evt.c, main/event.c,
include/asterisk/strings.h, main/asterisk.c,
channels/chan_mgcp.c, apps/app_voicemail.c,
channels/chan_unistim.c, include/asterisk/devicestate.h, /,
channels/chan_sip.c, main/devicestate.c: Merged revisions 184339
via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r184339 | russell | 2009-03-25 16:57:19 -0500 (Wed, 25
Mar 2009) | 35 lines Improve performance of the ast_event cache
functionality. This code comes from
svn/asterisk/team/russell/event_performance/. Here is a summary
of the changes that have been made, in order of both invasiveness
and performance impact, from smallest to largest. 1) Asterisk
1.6.1 introduces some additional logic to be able to handle
distributed device state. This functionality comes at a cost. One
relatively minor change in this patch is that the extra
processing required for distributed device state is now
completely bypassed if it's not needed. 2) One of the things that
I noticed when profiling this code was that a _lot_ of time was
spent doing string comparisons. I changed the way strings are
represented in an event to include a hash value at the front. So,
before doing a string comparison, we do an integer comparison on
the hash. 3) Finally, the code that handles the event cache has
been re-written. I tried to do this in a such a way that it had
minimal impact on the API. I did have to change one API call,
though - ast_event_queue_and_cache(). However, the way it works
now is nicer, IMO. Each type of event that can be cached (MWI,
device state) has its own hash table and rules for hashing and
comparing objects. This by far made the biggest impact on
performance. For additional details regarding this code and how
it was tested, please see the review request. (closes issue
#14738) Reported by: russell Review:
http://reviewboard.digium.com/r/205/ ........
2009-03-25 19:27 +0000 [r184266-184283] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 184280 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r184280 |
file | 2009-03-25 16:22:06 -0300 (Wed, 25 Mar 2009) | 5 lines Fix
issue with a T38 reinvite being sent even if not configured to do
so. If we receive a T38 request negotiate control frame we should
only attempt to do so if the option is enabled on the dialog.
........
* main/bridging.c, /: Merged revisions 183652 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r183652 |
file | 2009-03-22 18:00:28 -0300 (Sun, 22 Mar 2009) | 4 lines Fix
a minor logic flaw with the bridge generic thread. We only want
to move the channel pointers that are actually present. ........
2009-03-25 15:33 +0000 [r184256] Eliel C. Sardanons <eliels@gmail.com>
* main/asterisk.c, /: Merged revisions 184220 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r184220 | eliel | 2009-03-25 10:38:19 -0400 (Wed, 25 Mar 2009) |
19 lines Merged revisions 184188 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r184188 | eliel | 2009-03-25 10:12:54 -0400 (Wed, 25 Mar 2009) |
13 lines Avoid destroying the CLI line when moving the cursor
backward and trying to autocomplete. When moving the cursor
backward and pressing TAB to autocomplete, a NULL is put in the
line and we are loosing what we have already wrote after the
actual cursor position. (closes issue #14373) Reported by: eliel
Patches: asterisk.c.patch uploaded by eliel (license 64) Tested
by: lmadsen ........ ................
2009-03-25 14:40 +0000 [r184150-184221] Russell Bryant <russell@digium.com>
* main/timing.c, /: Merged revisions 184219 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r184219 |
russell | 2009-03-25 09:33:32 -0500 (Wed, 25 Mar 2009) | 2 lines
Include poll-compat.h ........
* main/timing.c, /: Merged revisions 184151 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r184151 |
russell | 2009-03-24 21:03:13 -0500 (Tue, 24 Mar 2009) | 2 lines
Change poll() to ast_poll(). ........
* utils/Makefile, /, include/asterisk/compat.h: Merged revisions
184147 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r184147 |
russell | 2009-03-24 20:42:10 -0500 (Tue, 24 Mar 2009) | 5 lines
Fix build issues on Mac OSX. (closes issue #14714) Reported by:
ygor ........
2009-03-24 22:42 +0000 [r184082] Mark Michelson <mmichelson@digium.com>
* apps/app_senddtmf.c, /: Merged revisions 184079 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r184079 | mmichelson | 2009-03-24 17:40:39 -0500 (Tue, 24 Mar
2009) | 15 lines Merged revisions 184078 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r184078 | mmichelson | 2009-03-24 17:34:45 -0500 (Tue, 24 Mar
2009) | 9 lines Change NULL pointer check to be ast_strlen_zero.
The 'digit' variable is guaranteed to be non-NULL, so the if
statement could never evaluate true. Changing to ast_strlen_zero
makes the logic correct. This was found while reviewing
ast_channel_ao2 code review. ........ ................
2009-03-24 22:02 +0000 [r184041-184044] Russell Bryant <russell@digium.com>
* main/channel.c, /: Merged revisions 184043 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r184043 |
russell | 2009-03-24 17:00:58 -0500 (Tue, 24 Mar 2009) | 2 lines
Put siren7 and siren14 in ast_best_codec() just so they're in
there somewhere. ........
* channels/chan_iax2.c, /: Merged revisions 184037 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r184037 | russell | 2009-03-24 16:40:44 -0500 (Tue, 24 Mar 2009)
| 6 lines Exclude slin16, siren7, and siren14 from bandwidth=low
and =medium The default codec configuration for chan_iax2 is
bandwidth=low. I noticed slin16 being negotiated as the codec in
some test calls, but that no longer happens after this change.
........
2009-03-24 15:29 +0000 [r183868-183917] Tilghman Lesher <tlesher@digium.com>
* /, configs/voicemail.conf.sample: Merged revisions 183914 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r183914 | tilghman | 2009-03-24 10:26:42 -0500
(Tue, 24 Mar 2009) | 10 lines Merged revisions 183913 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r183913 | tilghman | 2009-03-24 10:25:42 -0500 (Tue, 24 Mar 2009)
| 3 lines Additionally note that the operator option needs an 'o'
extension. (Related to issue #14731) ........ ................
* /, main/http.c: Merged revisions 183865 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r183865 |
tilghman | 2009-03-23 18:28:20 -0500 (Mon, 23 Mar 2009) | 2 lines
Allow browsers to cache images and other static content. (This is
a regression over 1.4) ........
2009-03-23 19:00 +0000 [r183769] Mark Michelson <mmichelson@digium.com>
* res/res_monitor.c, /: Merged revisions 183766 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r183766 | mmichelson | 2009-03-23 13:58:03 -0500 (Mon, 23 Mar
2009) | 13 lines Merged revisions 183700 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r183700 | mmichelson | 2009-03-23 12:59:28 -0500 (Mon, 23 Mar
2009) | 7 lines Fix a memory leak in res_monitor.c The only way
that this leak would occur is if Monitor were started using the
Manager interface and no File: header were given. Discovered
while reviewing the ast_channel_ao2 review request. ........
................
2009-03-23 18:12 +0000 [r183704] Leif Madsen <lmadsen@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 183701 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r183701 | lmadsen | 2009-03-23 14:06:40 -0400 (Mon, 23 Mar 2009)
| 7 lines Fixes a documentation error introduced during the CLI
cleanup at AstriDevCon 2008. (closes issue #14655) Reported by:
ulogic Patches: chan_dahdi.patch uploaded by ulogic (license 728)
Tested by: lmadsen ........
2009-03-20 17:09 +0000 [r183564] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c, /: Merged revisions 183560 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r183560 | russell | 2009-03-20 12:00:58 -0500
(Fri, 20 Mar 2009) | 10 lines Merged revisions 183559 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r183559 | russell | 2009-03-20 11:53:25 -0500 (Fri, 20 Mar 2009)
| 2 lines Fix a crash in IAX2 registration handling found during
load testing with dvossel. ........ ................
2009-03-20 12:19 +0000 [r183519] Eliel C. Sardanons <eliels@gmail.com>
* channels/chan_dahdi.c, /: Merged revisions 183511 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r183511 | eliel | 2009-03-20 08:12:49 -0400 (Fri, 20 Mar 2009) |
2 lines Remove duplicate <description> inside the xml
documentation. ........
2009-03-19 19:20 +0000 [r183337] Tilghman Lesher <tlesher@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 183321 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk
................ r183321 | tilghman | 2009-03-19 14:17:31 -0500
(Thu, 19 Mar 2009) | 15 lines Merged revisions 183319 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r183319 | tilghman | 2009-03-19 14:15:33 -0500 (Thu, 19 Mar 2009)
| 8 lines Delay signalling progress until a PRI channel really
signals progress. (closes issue #13034) Reported by: klaus3000
Patches: 20090316__bug13034.diff.txt uploaded by tilghman
(license 14) patch_trunk_183progress_klaus3000.txt uploaded by
klaus3000 (license 65) Tested by: klaus3000 ........
................
2009-03-19 18:20 +0000 [r183263] Russell Bryant <russell@digium.com>
* main/loader.c, /, configure, include/asterisk/autoconfig.h.in,
configure.ac: Merged revisions 183242 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r183242 | russell | 2009-03-19 13:00:15 -0500 (Thu, 19 Mar 2009)
| 10 lines Merged revisions 183241 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r183241 | russell | 2009-03-19 12:52:52 -0500 (Thu, 19 Mar 2009)
| 2 lines Remove the use of RTLD_NOLOAD, as it is not behaving
like expected. ........ ................
2009-03-19 18:12 +0000 [r183247] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c, /: Merged revisions 183244 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r183244 |
mmichelson | 2009-03-19 13:10:34 -0500 (Thu, 19 Mar 2009) | 16
lines Fix a memory leak associated with queues. For every attempt
that app_queue made to place an outbound call to a queue member,
we would allocate a queue_end_bridge structure. When the bridge
for the call had completed, we would free the structure.
Unfortunately not all call attempts actually end up bridged to a
member, so we need to be more selective of when to allocate the
structure. With this change, the allocation occurs in an area
where we can guarantee that the call will be bridged. (closes
issue #14680) Reported by: caspy Patches: 14680.patch uploaded by
mmichelson (license 60) Tested by: caspy ........
2009-03-19 Leif Madsen <lmadsen@digium.com>
* Release Asterisk 1.6.2.0-beta1
2009-03-19 16:11 +0000 [r183122] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 183117 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r183117 | mmichelson | 2009-03-19 11:07:54 -0500 (Thu, 19 Mar
2009) | 20 lines Merged revisions 183115 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r183115 | mmichelson | 2009-03-19 11:04:02 -0500 (Thu, 19 Mar
2009) | 14 lines Fix an issue where cancelled outgoing SIP calls
would erroneously report the device as "in use." A user was
having an issue where if an outgoing SIP call was canceled, the
SIP device would remain in use if we had not received any
response to the initial INVITE we sent out. The SIP device would
remain in use until the autocongestion timer was exhausted. I
tracked down the cause of this to be the section of code I am
removing here. I asked several people what the purpose of this
code was meant to be, but no one could give me any sort of answer
as to why this was here. The person who was having this issue has
been using this patch for several months and it has stopped the
problems they have had. AST-196 ........ ................
2009-03-19 15:45 +0000 [r183068-183111] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 183108 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r183108 |
file | 2009-03-19 12:37:23 -0300 (Thu, 19 Mar 2009) | 11 lines
Improve our triggering of a T38 switchover internally when
triggered by a received reinvite. Previously we reached across
the channel bridge to get the other party's SIP dialog structure
in order to trigger an outgoing reinvite. This is extremely
dangerous to do and only works if bridged to another SIP channel.
This patch changes this to use the T38 control frame method of
requesting a switchover. This change also causes the SIP channel
driver to propogate back whether the switchover worked or not
instead of blindly accepting the incoming T38 reinvite. Review:
http://reviewboard.digium.com/r/200/ ........
* main/channel.c, /: Merged revisions 183057 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r183057 |
file | 2009-03-18 19:22:56 -0300 (Wed, 18 Mar 2009) | 6 lines Fix
an issue where a T38 control frame would get dropped. If two
channels were bridged together using a generic bridge the T38
control frame would get passed up instead of being indicated on
the other channel. ........
2009-03-18 21:19 +0000 [r183031] Jeff Peeler <jpeeler@digium.com>
* /, channels/h323/ast_h323.cxx: Merged revisions 183028 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r183028 | jpeeler | 2009-03-18 16:18:27 -0500 (Wed, 18
Mar 2009) | 4 lines Add some code removed by mistake from commit
182722 that works around a file descriptor leak in versions of
PWLib prior to 1.12.0. ........
2009-03-18 14:39 +0000 [r182947] Russell Bryant <russell@digium.com>
* main/poll.c, main/io.c, main/channel.c, channels/chan_skinny.c,
configure, apps/app_mp3.c, res/res_agi.c,
include/asterisk/poll-compat.h, channels/chan_alsa.c,
main/asterisk.c, apps/app_nbscat.c, /, main/Makefile,
include/asterisk/autoconfig.h.in, configure.ac,
include/asterisk/io.h, main/utils.c, include/asterisk/channel.h:
Merged revisions 182847 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r182847 | russell | 2009-03-17 21:28:55 -0500 (Tue, 17 Mar 2009)
| 52 lines Merged revisions 182810 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009)
| 44 lines Fix cases where the internal poll() was not being used
when it needed to be. We have seen a number of problems caused by
poll() not working properly on Mac OSX. If you search around,
you'll find a number of references to using select() instead of
poll() to work around these issues. In Asterisk, we've had poll.c
which implements poll() using select() internally. However, we
were still getting reports of problems. vadim investigated a bit
and realized that at least on his system, even though we were
compiling in poll.o, the system poll() was still being used. So,
the primary purpose of this patch is to ensure that we're using
the internal poll() when we want it to be used. The changes are:
1) Remove logic for when internal poll should be used from the
Makefile. Instead, put it in the configure script. The logic in
the configure script is the same as it was in the Makefile.
Ideally, we would have a functionality test for the problem, but
that's not actually possible, since we would have to be able to
run an application on the _target_ system to test poll()
behavior. 2) Always include poll.o in the build, but it will be
empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll()
throughout the source tree to ast_poll(). I feel that it is good
practice to give the API call a new name when we are changing its
behavior and not using the system version directly in all cases.
So, normally, ast_poll() is just redefined to poll(). On systems
where AST_POLL_COMPAT is defined, ast_poll() is redefined to
ast_internal_poll(). 4) Change poll() in main/poll.c to be
ast_internal_poll(). It's worth noting that any code that still
uses poll() directly will work fine (if they worked fine before).
So, for example, out of tree modules that are using poll() will
not stop working or anything. However, for modules to work
properly on Mac OSX, ast_poll() needs to be used. (closes issue
#13404) Reported by: agalbraith Tested by: russell, vadim
http://reviewboard.digium.com/r/198/ ........ ................
2009-03-17 20:53 +0000 [r182725] Jeff Peeler <jpeeler@digium.com>
* channels/h323/chan_h323.h, channels/h323/compat_h323.cxx, /,
channels/h323/ast_h323.cxx, configure,
autoconf/ast_check_openh323.m4, channels/h323/compat_h323.h,
channels/chan_h323.c, channels/h323/ast_h323.h: Merged revisions
182722 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r182722 |
jpeeler | 2009-03-17 15:47:31 -0500 (Tue, 17 Mar 2009) | 15 lines
Allow H.323 Plus library to be used in addition to the OpenH323
library Chan_h323 can now be compiled against both the previously
supported versions of OpenH323 as well as the current H.323 Plus
(version 1.20.2). The configure script has been modified to look
in the default install location of h323 to hopefully help avoid
using the environment variables OPENH323DIR and PWLIBDIR. Also,
the CLI command "h323 show version" has been added which
indicates which version of h323 is in use. (closes issue #11261)
Reported by: vhatz Patches: asterisk-1.6.0.6-h323plus.patch
uploaded by jthurman (license 614) ........
2009-03-17 16:46 +0000 [r182592] Russell Bryant <russell@digium.com>
* main/channel.c, /: Merged revisions 182553 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r182553 |
russell | 2009-03-17 10:22:12 -0500 (Tue, 17 Mar 2009) | 5 lines
Tweak the handling of the frame list inside of ast_answer(). This
does not change any behavior, but moves the frames from the local
frame list back to the channel read queue using an O(n) algorithm
instead of O(n^2). ........
2009-03-17 15:01 +0000 [r182528-182534] Kevin P. Fleming <kpfleming@digium.com>
* main/channel.c, /: Merged revisions 182530 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r182530 |
kpfleming | 2009-03-17 09:59:33 -0500 (Tue, 17 Mar 2009) | 2
lines correct logic flaw in ast_answer() changes in r182525
........
* main/channel.c, /, main/features.c, include/asterisk/channel.h:
Merged revisions 182525 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r182525 |
kpfleming | 2009-03-17 09:38:11 -0500 (Tue, 17 Mar 2009) | 11
lines Improve behavior of ast_answer() to not lose incoming
frames ast_answer(), when supplied a delay before returning to
the caller, use ast_safe_sleep() to implement the delay.
Unfortunately during this time any incoming frames are discarded,
which is problematic for T.38 re-INVITES and other sorts of
channel operations. When a delay is not passed to ast_answer(),
it still delays for up to 500 milliseconds, waiting for media to
arrive. Again, though, it discards any control frames, or
non-voice media frames. This patch rectifies this situation, by
storing all incoming frames during the delay period on a list,
and then requeuing them onto the channel before returning to the
caller. http://reviewboard.digium.com/r/196/ ........
2009-03-17 05:54 +0000 [r182453] Tilghman Lesher <tlesher@digium.com>
* main/db.c, /: Merged revisions 182450 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r182450 | tilghman | 2009-03-17 00:51:54 -0500 (Tue, 17 Mar 2009)
| 14 lines Merged revisions 182449 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r182449 | tilghman | 2009-03-17 00:50:52 -0500 (Tue, 17 Mar 2009)
| 7 lines Fix race in astdb The underlying db1 implementation
does not fully isolate the pages retrieved from astdb, so the
lock protecting accesses needs to be extended until the copy from
the shared memory structure is done. (closes issue #14682)
Reported by: makoto ........ ................
2009-03-17 02:02 +0000 [r182409] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 182408 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r182408 | rmudgett | 2009-03-16 20:54:53 -0500 (Mon, 16 Mar 2009)
| 8 lines OPENR2 uses an incorrect string value if the extension
delimiter is not present. * Fixed OPENR2 using an incorrect
string value if the extension delimiter is not present in the
Dial() function. This was fixed for SS7 and PRI in trunk
-r172400. * Made OPENR2 stripmsd behavior the same as the SS7,
PRI, and others. * Removed trailing whitespace that appeared with
OPENR2. ........
2009-03-16 20:51 +0000 [r182360-182361] Russell Bryant <russell@digium.com>
* /: svnmerge init
* / (added): Create a branch for 1.6.2
2009-03-16 20:35 +0000 [r182355] Russell Bryant <russell@digium.com>
* CREDITS, channels/chan_dahdi.c, configs/chan_dahdi.conf.sample,
configure, include/asterisk/autoconfig.h.in, configure.ac,
CHANGES, makeopts.in: Add MFC/R2 support for chan_dahdi. This
commit introduces official support for R2 signaling in
chan_dahdi. The modifications to chan_dahdi, and the supporting
library, LibOpenR2, were both written by Moises Silva. Many users
are using this code, or a variant of it, in Asterisk 1.2, 1.4 and
1.6 in Brazil, México and Argentina. An unknown number of users
(but at least 1) are using it in each of the following countries:
Colombia, Nepal, Thailand, Venezuela, Perú, and probably others.
To use this code, LibOpenR2 must be installed from
http://www.libopenr2.org/. Information about configuration can be
found in configs/chan_dahdi.conf.sample. The code committed is
the most up to date version, which was being maintained in
svn/asterisk/team/moy/mfcr2/. I would also like to include a
Thank You to the many others that tested this code beyond those
listed in this commit message. These are the names that I could
find in the mantis issue. (closes issue #12509) Reported by: moy
Patches: chan_zap-mfr2.patch uploaded by moy (license 222) Tested
by: moy, korihor, viniciusfontes, Skarmeth, loloski,
asbestoshead, titogarrido, heliocoelhojr, konsultex, ncorrare,
ecarruda, rtorresduque, PTorres, ychen Review:
http://reviewboard.digium.com/r/40/
2009-03-16 17:49 +0000 [r182282] David Vossel <dvossel@digium.com>
* /, channels/chan_iax2.c: Merged revisions 182281 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r182281 | dvossel | 2009-03-16 12:47:42 -0500 (Mon, 16
Mar 2009) | 7 lines Randomize IAX2 encryption padding The 16-32
byte random padding at the beginning of an encrypted IAX2 frame
turns out to not be all that random at all. This patch calls
ast_random to fill the padding buffer with random data. The
padding is randomized at the beginning of every encrypted call
and for every encrypted retransmit frame. Review:
http://reviewboard.digium.com/r/193/ ........
2009-03-16 17:33 +0000 [r182211-182278] Tilghman Lesher <tlesher@digium.com>
* funcs/func_env.c: Fix an off-by-one error in the FILE() function,
and extend FILE()'s length parameter to work like variable
substitution. Previously, FILE() returned one less character than
specified, due to the terminating NULL. Both the offset and
length parameters now behave identically to the way variable
substitution offsets and lengths also work. (closes issue #14670)
Reported by: BMC
* channels/chan_local.c, /: Merged revisions 182208 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r182208 | tilghman | 2009-03-16 10:39:15 -0500 (Mon, 16
Mar 2009) | 7 lines Fixup glare detection, to fix a memory leak
of a local pvt structure. (closes issue #14656) Reported by:
caspy Patches: 20090313__bug14656__2.diff.txt uploaded by
tilghman (license 14) Tested by: caspy ........
2009-03-16 13:58 +0000 [r182171] Joshua Colp <jcolp@digium.com>
* main/channel.c: Fix a memory leak in the ast_answer /
__ast_answer API call. For a channel that is not yet answered
this API call will wait until a voice frame is received on the
channel before returning. It does this by waiting for frames on
the channel and reading them in. The frames read in were not
freed when they should have been.
2009-03-13 21:26 +0000 [r182029-182121] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Change faulty comparison used when announcing
average hold minutes and seconds (closes issue #14227) Reported
by: caspy
* main/features.c: Remove ast_ prefix from functions which are not
public.
* /, main/features.c: Merged revisions 181990 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r181990 | mmichelson | 2009-03-13 12:12:32 -0500 (Fri, 13 Mar
2009) | 35 lines Check the DYNAMIC_FEATURES of both the chan and
peer when interpreting DTMF. Dynamic features defined in the
applicationmap section of features.conf allow one to specify
whether the caller, callee, or both have the ability to use the
feature. The documentation in the features.conf.sample file could
be interpreted to mean that one only needs to set the
DYNAMIC_FEATURES channel variable on the calling channel in order
to allow for the callee to be able to use the features which he
should have permission to use. However, the DYNAMIC_FEATURES
variable would only be read from the channel of the participant
that pressed the DTMF sequence to activate the feature. The
result of this was that the callee was unable to use dynamic
features unless the dialplan writer had taken measures to be sure
that the DYNAMIC_FEATURES variable was set on the callee's
channel. This commit changes the behavior of
ast_feature_interpret to concatenate the values of
DYNAMIC_FEATURES from both parties involved in the bridge. The
features themselves determine who has permission to use them, so
there is no reason to believe that one side of the bridge could
gain the ability to perform an action that they should not have
the ability to perform. Kevin Fleming pointed out on the
asterisk-users list that the typical way that this was worked
around in the past was by setting _DYNAMIC_FEATURES on the
calling channel so that the value would be inherited by the
called channel. While this works, the documentation alone is not
enough to figure out why this is necessary for the callee to be
able to use dynamic features. In this particular case, changing
the code to match the documentation is safe, easy, and will
generally make things easier for people for future installations.
This bug was originally reported on the asterisk-users list by
David Ruggles. (closes issue #14657) Reported by: mmichelson
Patches: 14657.patch uploaded by mmichelson (license 60) ........
2009-03-13 17:25 +0000 [r182022] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Fix an issue with requesting a T38 reinvite
before the call is answered. The code responsible for sending the
T38 reinvite did not check if an INVITE was already being
handled. This caused things to get confused and the call to fail.
The code now defers sending the T38 reinvite until the current
INVITE is done being handled. (issue AST-191)
2009-03-13 16:55 +0000 [r181985] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_sip.c: improve a bit of suboptimal code
2009-03-13 01:26 +0000 [r181899] Richard Mudgett <rmudgett@digium.com>
* /: Merged revisions 181898 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 Just
recording the v1.4 change in trunk since it originally came from
here. ........ r181898 | rmudgett | 2009-03-12 20:19:29 -0500
(Thu, 12 Mar 2009) | 4 lines Use the correct branch integrated
property when generating the version string. Copied the
make_version file from Asterisk trunk. ........
2009-03-12 21:43 +0000 [r181769-181846] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Run the macro on the queue member's channel
when he answers, not the caller's channel.
* /, channels/chan_sip.c: Merged revisions 181768 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r181768 | mmichelson | 2009-03-12 13:29:48 -0500 (Thu, 12 Mar
2009) | 22 lines Properly send a 487 on an INVITE we have not
responded to if we receive a BYE. If we receive an INVITE from an
endpoint and then later receive a BYE from that same endpoint
before we have sent a final response for the INVITE, then we need
to respond to the INVITE with a 487. There was logic in the code
prior to this commit which seemed to exist solely to handle this
situation, but there was one condition in an if statement which
was incorrect. The only way we would send a 487 was if the
sip_pvt had no owner channel. This made no sense since we created
the owner channel when we received the INVITE, meaning that the
majority of the time we would never send the 487. The 487 being
sent should not rely on whether we have created a channel. Its
delivery should be dependent on the current state of the initial
INVITE transaction. With this commit, that logic is now correctly
in place. (closes issue #14149) Reported by: legranjl Patches:
14149.patch uploaded by mmichelson (license 60) Tested by:
legranjl ........
2009-03-12 17:32 +0000 [r181731] Tilghman Lesher <tlesher@digium.com>
* main/translate.c: Adjust translation table column widths based
upon the translation times. Previously, only 5 columns were
displayed, and if a translation time exceeded 99,999 useconds, it
would be displayed as 0, instead of its actual time. (closes
issue #14532) Reported by: pj Patches:
20090311__bug14532.diff.txt uploaded by tilghman (license 14)
Tested by: pj
2009-03-12 16:56 +0000 [r181612-181665] Joshua Colp <jcolp@digium.com>
* /, res/res_musiconhold.c: Merged revisions 181664 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r181664 | file | 2009-03-12 13:56:20 -0300 (Thu, 12 Mar
2009) | 2 lines Fix incorrect usage of strncasecmp... I really
meant to use strcasecmp. ........
* /, res/res_musiconhold.c: Merged revisions 181659-181660 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r181659 | file | 2009-03-12 13:50:37 -0300 (Thu, 12 Mar 2009) | 8
lines Fix another scenario where depending on configuration the
stream would not get read. For custom commands we don't know
whether the audio is coming from a stream or not so we are going
to have to read the data despite no channels. (closes issue
#14416) Reported by: caspy ........ r181660 | file | 2009-03-12
13:52:45 -0300 (Thu, 12 Mar 2009) | 2 lines Fix logic flaw in
previous commit. ........
* /, res/res_musiconhold.c: Merged revisions 181655 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r181655 | file | 2009-03-12 13:29:19 -0300 (Thu, 12 Mar
2009) | 10 lines Fix issue with streaming MOH failing if nobody
is listening. When a music class is setup to actually provide
music on hold from a stream we need to constantly read audio from
it since it will constantly be providing audio. This is now done
despite there being no channels listening to it. (closes issue
#14416) Reported by: caspy ........
* apps/app_dial.c: Fix crash when sleep and retries argument was
not given to RetryDial application. (closes issue #14647)
Reported by: sherpya
2009-03-12 01:33 +0000 [r181542-181577] Richard Mudgett <rmudgett@digium.com>
* build_tools/make_version: Whitespace chages.
* build_tools/make_version: Use the correct branch integrated
property when generating the version string
2009-03-11 23:14 +0000 [r181499] Michiel van Baak <michiel@vanbaak.info>
* configs/sip.conf.sample: Provide correct hint to debug SIP
trouble in the default config (closes issue #14646) Reported by:
strk
2009-03-11 22:25 +0000 [r181465] Russell Bryant <russell@digium.com>
* main/channel.c: Make handling of the BRIDGE_PLAY_SOUND variable
thread-safe.
2009-03-11 22:20 +0000 [r181444] Jason Parker <jparker@digium.com>
* /, configure, configure.ac: Merged revisions 181436 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r181436 | qwell | 2009-03-11 17:18:42 -0500 (Wed, 11 Mar
2009) | 4 lines Allow prefix to set localstatedir (when used and
different from the default). This is similar to the /etc change
that was made for the non-FreeBSD case. ........
2009-03-11 22:14 +0000 [r181424-181428] Russell Bryant <russell@digium.com>
* main/channel.c: Make handling of the BRIDGEPVTCALLID variable
thread-safe.
* main/channel.c, /: Merged revisions 181423 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r181423 | russell | 2009-03-11 16:42:58 -0500 (Wed, 11 Mar 2009)
| 9 lines Make code that updates BRIDGEPEER variable thread-safe.
It is not safe to read the name field of an ast_channel without
the channel locked. This patch fixes some places in channel.c
where this was being done, and lead to crashes related to
masquerades. (closes issue #14623) Reported by: guillecabeza
........
2009-03-11 17:34 +0000 [r181371] David Vossel <dvossel@digium.com>
* channels/iax2-parser.h, /, channels/chan_iax2.c: Merged revisions
181340 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r181340 | dvossel | 2009-03-11 12:25:31 -0500 (Wed, 11 Mar 2009)
| 11 lines encrypted IAX2 during packet loss causes decryption to
fail on retransmitted frames If an iax channel is encrypted, and
a retransmit frame is sent, that packet's iseqno is updated while
it is encrypted. This causes the entire frame to be corrupted.
When the corrupted frame is sent, the other side decrypts it and
sends a VNAK back because the decrypted frame doesn't make any
sense. When we get the VNAK, we look through the sent queue and
send the same corrupted frame causing a loop. To fix this,
encrypted frames requiring retransmission are decrypted, updated,
then re-encrypted. Since key-rotation may change the key held by
the pvt struct, the keys used for encryption/decryption are held
within the iax_frame to guarantee they remain correct. (closes
issue #14607) Reported by: stevenla Tested by: dvossel Review:
http://reviewboard.digium.com/r/192/ ........
2009-03-11 17:26 +0000 [r181345] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 181328 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r181328 | file | 2009-03-11 14:22:52 -0300 (Wed, 11 Mar 2009) |
14 lines Fix issue where an attended transfer could not be
completed under a rare scenario. When completing an attended
transfer chan_sip does a check to make sure the extension in the
URI portion of the Refer-To header is a local valid extension. We
don't actually need to check this since we know for sure the
other channel is already up and talking to the extension. Some
devices do not put the extension in the Refer-To header either,
which can cause the extension check to fail. We now no longer do
this check if it is an attended transfer. (closes issue #14628)
Reported by: sverre Patches: 14628.diff uploaded by file (license
11) ........
2009-03-11 17:04 +0000 [r181301] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/astobj2.h: Turn off malloc debugging of astobj2,
since it apparently doesn't work too well during startup.
2009-03-11 16:40 +0000 [r181296] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 181295 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r181295 | file | 2009-03-11 13:36:50 -0300 (Wed, 11 Mar 2009) | 9
lines Fix a problem with inband DTMF detection on outgoing SIP
calls when dtmfmode=auto. When dtmfmode was set to auto the
inband DTMF detector was not setup on outgoing SIP calls. This
caused inband DTMF detection to fail. The inband DTMF detector is
now setup for both dtmfmode inband and auto. (closes issue
#13713) Reported by: makoto ........
2009-03-11 16:19 +0000 [r181292] Russell Bryant <russell@digium.com>
* doc/google-soc2009-ideas.txt: Replace contents of this doc with a
pointer to its new home
2009-03-11 14:28 +0000 [r181244] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Fix segfault when dialing a typo'd queue If
trying to dial a non-existent queue, there would be a segfault
when attempting to access q->weight, even though q was NULL. This
problem was introduced during the queue-reset merge and thus only
affects trunk. (closes issue #14643) Reported by: alecdavis
2009-03-11 13:44 +0000 [r181210] Joshua Colp <jcolp@digium.com>
* apps/app_confbridge.c: Don't play the "you are about to be placed
into the conference" and "the leader has left the conference"
sounds if the quiet option is enabled. (reported by Vadim Lebedev
on the asterisk-dev list)
2009-03-11 04:06 +0000 [r181135] Jeff Peeler <jpeeler@digium.com>
* utils/Makefile, include/asterisk/utils.h,
include/asterisk/astmm.h, channels/chan_sip.c,
channels/h323/ast_h323.cxx, main/features.c, utils/extconf.c,
pbx/pbx_config.c: Fix malloc debug macros to work properly with
h323. The main problem here was that cstdlib was undefining free
thereby causing the proper debug macros to not be used.
ast_h323.cxx has been changed to call ast_free instead to avoid
the issue. A few other issues were addressed: - There were a few
instances of functions improperly passing ast_free instead of
ast_free_ptr. - Some clean up was done to avoid the debug macros
intentionally being redefined. (copied below from Kevin's commit,
appreciate the help) - disable astmm.h from doing anything when
STANDALONE is defined, which is used by the tools in the utils/
directory that use parts of Asterisk header files in hackish
ways; also ensure that utils/extconf.c and utils/conf2ael.c are
compiled with STANDALONE defined. (closes issue #13593) Reported
by: pj
2009-03-11 02:25 +0000 [r181099] Russell Bryant <russell@digium.com>
* doc/google-soc2009-ideas.txt: tabs to spaces
2009-03-11 00:49 +0000 [r181032-181033] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Add missing comment that quotes RFC 3891
* /, channels/chan_sip.c: Merged revisions 181029,181031 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r181029 | mmichelson | 2009-03-10 19:30:26 -0500 (Tue, 10 Mar
2009) | 9 lines Fix incorrect tag checking on transfers when
pedantic=yes is enabled. (closes issue #14611) Reported by:
klaus3000 Patches: patch_chan_sip_attended_transfer_1.4.23.txt
uploaded by klaus3000 (license 65) Tested by: klaus3000 ........
r181031 | mmichelson | 2009-03-10 19:32:40 -0500 (Tue, 10 Mar
2009) | 3 lines Remove unused variables. ........
2009-03-11 00:29 +0000 [r181027-181028] Tilghman Lesher <tlesher@digium.com>
* main/strings.c, main/hashtab.c, include/asterisk/astobj2.h,
main/heap.c, include/asterisk/strings.h,
include/asterisk/hashtab.h, main/astobj2.c,
include/asterisk/heap.h: Add MALLOC_DEBUG to various utility
APIs, so that memory leaks can be tracked back to their source.
(related to issue #14636)
* main/pbx.c: Spacing changes only
2009-03-10 22:03 +0000 [r180944] Jason Parker <jparker@digium.com>
* /, configure, configure.ac, autoconf/ast_prog_sed.m4,
autoconf/ast_check_gnu_make.m4: Merged revisions 180941 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r180941 | qwell | 2009-03-10 17:02:18 -0500 (Tue, 10 Mar 2009) |
1 line Make things happier when using autoconf 2.62+ ........
2009-03-10 22:03 +0000 [r180935-180942] Russell Bryant <russell@digium.com>
* doc/google-soc2009-ideas.txt: Add some notes on getting in
contact with the dev community
* doc/google-soc2009-ideas.txt: Remove difficulty and language
specifiers
* doc/google-soc2009-ideas.txt: Expand upon documentation of
manager event project
2009-03-10 21:15 +0000 [r180898] Michiel van Baak <michiel@vanbaak.info>
* CHANGES: list the move of the astvarrundir from /var/run to
/var/run/asterisk (actually its $(localstatedir)/run/asterisk
Makes setups with asterisk as non-root easier to manage because
you can setup permissions on this dir instead of touching a file
and setting permissions on that. Files that come to mind are
asterisk.pid and asterisk.ctl socket. Prodded by and ok @russell
2009-03-10 19:36 +0000 [r180859-180862] Russell Bryant <russell@digium.com>
* doc/google-soc2009-ideas.txt: add more projects
* doc/google-soc2009-ideas.txt: add more project ideas
2009-03-10 14:40 +0000 [r180800] Joshua Colp <jcolp@digium.com>
* main/manager.c: Reset the thread local string buffer when
handling the UserEvent action. (closes issue #14593) Reported by:
JimDickenson
2009-03-09 22:00 +0000 [r180750] Russell Bryant <russell@digium.com>
* doc/google-soc2009-ideas.txt: Add current mentors list, and first
pass on a project list broken out of "PineMango" I will work on
adding projects that have been sent to be via email tomorrow.
2009-03-09 20:58 +0000 [r180719] Jeff Peeler <jpeeler@digium.com>
* include/asterisk/rtp.h, include/asterisk/extconf.h,
main/devicestate.c, include/asterisk/tcptls.h, main/enum.c,
include/asterisk/callerid.h, include/asterisk/doxyref.h,
include/asterisk/event.h, include/asterisk/audiohook.h,
include/asterisk/dsp.h, include/asterisk/timing.h,
include/asterisk/udptl.h, include/asterisk/dlinkedlists.h,
include/asterisk/utils.h, include/asterisk/devicestate.h,
include/asterisk/taskprocessor.h, include/asterisk/enum.h,
include/asterisk/astobj2.h, include/asterisk/config.h,
include/asterisk/channel.h, include/asterisk/manager.h,
include/asterisk/heap.h, include/asterisk/logger.h,
include/asterisk/http.h, include/asterisk/res_odbc.h,
include/asterisk/app.h, main/tcptls.c,
include/asterisk/linkedlists.h, include/asterisk/sched.h,
include/asterisk/datastore.h, include/asterisk/lock.h,
include/asterisk/pbx.h, include/asterisk/dnsmgr.h: Add Doxygen
documentation for API changes from 1.6.0 to 1.6.1 Copied from my
review board description: This is a continuation of the API
changes documentation started for describing changes between
releases. Most of the API changes were pretty simple needing only
to be brought to attention via the new "Asterisk API Changes"
list. However, if you see anything that needs further explanation
feel free to supplement what is there. The current method of
documenting is to add (in the header file): \version <ver number>
<description of changes> and then to add the function to the
change list in doxyref.h on the AstAPIChanges page. I also made
sure all the functions that were newly added were tagged with
\since 1.6.1. I think this is a good habit to start both for the
historical aspect as well as for the future ability to easily add
a "New Asterisk API" page. Review:
http://reviewboard.digium.com/r/190/
2009-03-09 14:14 +0000 [r180684] Russell Bryant <russell@digium.com>
* doc/google-soc2009-ideas.txt (added): Add skeleton for GSoC ideas
list
2009-03-07 15:36 +0000 [r180641] Russell Bryant <russell@digium.com>
* contrib/asterisk-ng-doxygen: Make some minor updates to the
doxygen configuration - add bridges directory to be processed -
add some res/ subdirs - alphabetize subdirs - use consistent
indentation
2009-03-06 18:25 +0000 [r180579] Mark Michelson <mmichelson@digium.com>
* /, apps/app_voicemail.c: Merged revisions 180567 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r180567 | mmichelson | 2009-03-06 12:23:09 -0600 (Fri,
06 Mar 2009) | 2 lines Make compilation succeed in dev-mode when
IMAP storage is enabled. ........
2009-03-06 17:26 +0000 [r180534] David Vossel <dvossel@digium.com>
* /, main/enum.c: Merged revisions 180532 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r180532 | dvossel | 2009-03-06 11:19:55 -0600 (Fri, 06 Mar 2009)
| 9 lines Fix handling of backreferences for ENUM lookups enum.c
did not handle regex backtraces correctly. The '\1' in the regex
is a backreference that requires a pattern match to be inserted.
The way the code used to work is that it would find the
backreference and insert the entire input string minus the '+'.
This is incorrect. The regexec() function takes in a variable
called pmatch which is an array of structs containing the start
and end indexes for each backreference substring. The original
code actually passed the pmatch array pointer into regexec but
never did anything with it. Now when a backtrace is found, the
backtrace number is looked up in the pmatch array and the correct
substring is inserted. (closes issue #14576) Reported by:
chris-mac Review: http://reviewboard.digium.com/r/187/ ........
2009-03-05 23:26 +0000 [r180383-180465] Mark Michelson <mmichelson@digium.com>
* /, apps/app_voicemail.c: Merged revisions 180464 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r180464 | mmichelson | 2009-03-05 17:26:11 -0600 (Thu,
05 Mar 2009) | 16 lines [IMAP] Fix message retrieval issues when
identical mailbox names were defined in separate contexts. There
was a fix put in a while back so that an X-Asterisk-VM-Context
message header was added to stored IMAP voicemails. This would
allow for us to differentiate if the same mailbox name was used
in multiple contexts. The problem still left was that not all
places where messages were retrieved actually attempted to use
this header for information when retrieving messages. This commit
fixes that so that MWI and message retrieval from VoiceMailMain
work as expected. (closes issue #13853) Reported by: vicks1
Patches: 13853_v2.patch uploaded by mmichelson (license 60)
Tested by: lmadsen ........
* /, configs/voicemail.conf.sample, apps/app_voicemail.c: Merged
revisions 180380 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r180380 | mmichelson | 2009-03-05 12:58:48 -0600 (Thu, 05 Mar
2009) | 25 lines Fix broken mailbox parsing when searchcontexts
option is enabled. When using the searchcontexts option in
voicemail.conf, the code made the assumption that all mailbox
names defined were unique across all contexts. However, the code
did nothing to actually enforce this assumption, nor did it do
anything to alert a user that he may have created an ambiguity in
his voicemail.conf file by defining the same mailbox name in
multiple contexts. With this change, we now will issue a nice
long warning if searchcontexts is on and we encounter the same
mailbox name in multiple contexts and ignore any duplicates after
the first box. Whether searchcontexts is enabled or not, if we
come across a duplicate mailbox in the same context, then we will
issue a warning and ignore the duplicated mailbox. I have also
added a small note to voicemail.conf.sample in the explanation
for searchcontexts explaining that you cannot define the same
mailbox in multiple contexts if you have enabled the option.
(closes issue #14599) Reported by: lmadsen Patches: 14599.patch
uploaded by mmichelson (license 60) (with slight modification)
Tested by: lmadsen ........
2009-03-05 19:05 +0000 [r180382] Michiel van Baak <michiel@vanbaak.info>
* Makefile: Make sure we terminate the first s| command so we can
actually produce correct files.
2009-03-05 18:29 +0000 [r180373] Kevin P. Fleming <kpfleming@digium.com>
* main/frame.c, /, include/asterisk/frame.h, main/rtp.c: Merged
revisions 180372 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r180372 | kpfleming | 2009-03-05 12:22:16 -0600 (Thu, 05 Mar
2009) | 9 lines Fix problems when RTP packet frame size is
changed During some code analysis, I found that calling
ast_rtp_codec_setpref() on an ast_rtp session does not work as
expected; it does not adjust the smoother that may on the RTP
session, in fact it summarily drops it, even if it has data in
it, even if the current format's framing size has not changed.
This is not good. This patch changes this behavior, so that if
the packetization size for the current format changes, any
existing smoother is safely updated to use the new size, and if
no smoother was present, one is created. A new API call for
smoothers, ast_smoother_reconfigure(), was required to implement
these changes. Review: http://reviewboard.digium.com/r/184/
........
2009-03-05 18:18 +0000 [r180369] Joshua Colp <jcolp@digium.com>
* channels/chan_bridge.c (added), main/Makefile,
bridges/bridge_simple.c, bridges/bridge_softmix.c,
include/asterisk/channel.h, bridges/bridge_multiplexed.c,
CHANGES, Makefile, include/asterisk/bridging_technology.h
(added), bridges (added), bridges/bridge_builtin_features.c,
include/asterisk/bridging_features.h (added),
include/asterisk/bridging.h (added), apps/app_confbridge.c
(added), main/bridging.c (added), bridges/Makefile: Merge phase 1
support for the new bridging architecture. This commit brings in
the bridging core, bridging technologies, and the ConfBridge
application. For usage information on the ConfBridge application
please see the output of "core show application ConfBridge" from
the CLI. For API documentation please see the doxygen page
describing the architecture and the documentation for each API
call. Review: http://reviewboard.digium.com/r/93/
2009-03-05 06:21 +0000 [r180304-180334] Tilghman Lesher <tlesher@digium.com>
* contrib/editors/asterisk.vim: Also highlight the preamble and
postamble
* contrib/editors/ael.vim (added), contrib/editors/asterisk.vim
(added), contrib/editors (added), contrib/editors/asteriskvm.vim
(added): Add syntax coloring files for Vim, including a new one
for AEL
2009-03-04 21:01 +0000 [r180261] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Resolve object matching issues related to
the removal of the sip_user object. Previously, chan_sip had both
sip_peer and sip_user objects in memory. A patch went in to
remove sip_user to simplify the code, since everything could be
done with just sip_peer. This patch resolves some regressions
found that were introduced by those changes. This code comes from
svn/asterisk/team/group/sip-object-matching/. Here is a list of
the changes that have been made: 1) When doing a match by name
with the find_peer() function, make it much easier to specify
which objects should be matched by having a parameter that
specifies exactly which object types should be considered. Also,
update find_by_name() to handle this parameter. Finally, update
all code to use the new option values. 2) When looking up an
object for an outbound request by name, consider peers only.
(create_addr()) 3) Only match peers on an incoming registration
request. 4) When doing authentication (except for SUBSCRIBE),
look up users by name, instead of all objects by name. 5) When
doing authentication (except for SUBSCRIBE), after looking for a
user by name, look for a peer by IP address, instead of all
objects by IP address. 6) When handling the SIP qualify CLI
command or manager action, look for a peer by name, instead of
any object by name. 7) When handling the SIP unregister CLI
command, look for a peer by name, instead of any object by name.
9) In sip_do_debug_peer(), search for a peer by name, instead of
any object by name. 9) When handling the SIPPEER() dialplan
function, search for a peer by name, instead of any object by
name. 10) In the following session timer related functions,
st_get_se(), st_get_refresher(), and st_get_mode(), when looking
for an object for a given sip_pvt using pvt->peername, look for a
peer by name, instead of any object by name. 11) Fix build_peer()
to properly handle the case where separate type=peer and
type=user entries were specified in sip.conf. (closes issue
#14505) Reported by: lmadsen Review:
http://reviewboard.digium.com/r/172/
2009-03-04 20:48 +0000 [r180259] Tilghman Lesher <tlesher@digium.com>
* main/aescrypt.c, main/abstract_jb.c, main/acl.c, main/app.c,
main/alaw.c: Spacing changes only
2009-03-04 19:24 +0000 [r180195] Joshua Colp <jcolp@digium.com>
* /, main/callerid.c: Merged revisions 180194 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r180194 | file | 2009-03-04 15:22:50 -0400 (Wed, 04 Mar 2009) | 4
lines Look for the number in a callerid string starting from the
end. This way a value using <> can exist in the name portion.
(issue #AST-194) ........
2009-03-04 17:03 +0000 [r180155] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c, configs/sip.conf.sample: Allow for "magic"
pickups to work when we wish to ignore the context When the
subscription context for a call pickup subscription differs from
the context of the call pickup target, there's not an easy way to
divine what context should be used for the pickup. The way to
work around this is to use PICKUPMARK as the context for the
pickup. This has been documented in the sip.conf.sample file
(ABE-1708) closes issue #14567 submitted by: alecdavis
2009-03-04 14:39 +0000 [r180120] Joshua Colp <jcolp@digium.com>
* apps/app_dial.c: Remove duplicate 'k' and 'K' Dial options.
(closes issue #14601) Reported by: alecdavis Patches:
app_dial.optionk.diff.txt uploaded by alecdavis (license 585)
2009-03-03 23:35 +0000 [r180079] Steve Murphy <murf@digium.com>
* utils/Makefile: My bad! left check_expr2 in the ALL_UTILS list by
mistake. Already done to 1.6.x
2009-03-03 23:21 +0000 [r180032] David Vossel <dvossel@digium.com>
* main/channel.c, include/asterisk/app.h, apps/app_read.c,
main/app.c: app_read does not break from prompt loop with user
terminated empty string In app.c, ast_app_getdata is called to
stream the prompts and receive DTMF input. If ast_app_getdata()
receives an empty string caused by the user inputing the end of
string character, in this case '#', it should break from the
prompt loop and return to app_read, but instead it cycles through
all the prompts. I've added a return value for this special case
in ast_readstring() which uses an enum I've delcared in apps.h.
This enum is now used as a return value for ast_app_getdata().
(closes issue #14279) Reported by: Marquis Patches:
fix_app_read.patch uploaded by Marquis (license 32)
read-ampersanmd.patch2 uploaded by dvossel (license 671) Tested
by: Marquis, dvossel Review: http://reviewboard.digium.com/r/177/
2009-03-03 22:49 +0000 [r180007] Mark Michelson <mmichelson@digium.com>
* /, configs/queues.conf.sample, apps/app_queue.c: Merged revisions
180006 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r180006 | mmichelson | 2009-03-03 16:48:18 -0600 (Tue, 03 Mar
2009) | 17 lines Clarify some documentation of queues.conf.sample
It had always been possible to explicitly specify a "blank" value
for a sound file in queues.conf and have no sound played back.
The problem with this is that it would result in some ugly CLI
warnings from file.c. This commit introduces a check when playing
a file in app_queue to see if the name of the file is zero-length
and return early if that is the case. Also, the ability to
specify the blank sound files in queues.conf is now mentioned
more clearly in queues.conf.sample (closes issue #14227) Reported
by: caspy ........
2009-03-03 22:12 +0000 [r179973] Steve Murphy <murf@digium.com>
* utils/Makefile, utils/expr2.testinput, /, main/ast_expr2.h,
main/ast_expr2.y, main/ast_expr2f.c, main/ast_expr2.fl,
main/ast_expr2.c: Merged revisions 179807 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 I had some
work to do to port these changes to trunk; the check_expr stuff
hasn't been updated here for quite some time, it appears. I added
some more tests to the check_expr2 suite. I had to play around
with the makefile a bit, etc. I added STANDALONE2 #ifdefs to
ast_expr2.y so as not to conflict structure with aelparse.
........ r179807 | murf | 2009-03-03 11:11:34 -0700 (Tue, 03 Mar
2009) | 19 lines These changes allow AEL to better check ${}
constructs within $[...], that are concatenated with text. I
modified and added rules in ast_expr2.fl to better handle the
concatenations. I added some default routines to ast_expr2.y so
the standalone would compile. It also looks like I haven't run
this thru bison since 2.1, so it's good to get this updated. The
Makefile has comments added now for check_expr2 and check_expr to
explain what they are for, and how to run them. The testexpr2s
stuff has been removed, in favor of check_expr2. expr2.testinput
has been updated to include the two expressions that inspired
these changes (from mcnobody on #asterisk this morning) The
regression has been run and all looks well. ........
2009-03-03 22:01 +0000 [r179972] David Vossel <dvossel@digium.com>
* apps/app_meetme.c: app_meetme not setting filename and fileformat
correctly for realtime When app_meetme finds a realtime
conference, it doesn't get the filename and fileformat correctly
when 'r' is set. Now app_meetme first checks to see if fileformat
and filename are declared in the db, if they're not it checks the
.conf file, if its not declared there either it then uses
defaults. (closes issue #14545) Reported by: dalbaech Patches:
app_meetme-realtime5.patch uploaded by dvossel (license 671)
Realtime_Conference_Record_workaround.txt uploaded by dalbaech
(license 705) Tested by: dvossel, dalbaech Review:
http://reviewboard.digium.com/r/180/
2009-03-03 20:59 +0000 [r179937] Mark Michelson <mmichelson@digium.com>
* res/res_timing_dahdi.c, doc/timing.txt (added): Add documentation
for timing modules used in Asterisk This document specifies the
timing modules available in Asterisk beginning with Asterisk
1.6.1. The document goes into detail about the differences
between each and gives a general overview of what timing is used
for in Asterisk. There is also a section which can be used to
help customize your setup or to troubleshoot timing issues you
may have. I also added messages to the DAHDI timing test used in
res_timing_dahdi.c that points to this new documentation if
people experience problems. Big thanks to all who contributed
comments on this. (closes issue #14490) Reported by: mmichelson
Patches: timing.txt uploaded by mmichelson (license 60) Review:
http://reviewboard.digium.com/r/164/
2009-03-03 20:02 +0000 [r179903] Brian Degenhardt <bmd@digium.com>
* apps/app_directed_pickup.c: fix a leaked channel lock (and future
deadlock) when we try to pick up our own channel
2009-03-03 18:28 +0000 [r179841] Joshua Colp <jcolp@digium.com>
* /, main/features.c: Merged revisions 179840 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r179840 | file | 2009-03-03 14:27:09 -0400 (Tue, 03 Mar 2009) | 9
lines Do not assume that the bridge_cdr is still attached to the
channel when the 'h' exten is finished executing. It is possible
for a masquerade operation to occur when the 'h' exten is
operating. This operation moves the CDR records around causing
the bridge_cdr to no longer exist on the channel where it is
expected to. We can not safely modify it afterwards because of
this, so don't even try. (closes issue #14564) Reported by: meric
........
2009-03-03 17:03 +0000 [r179745] Mark Michelson <mmichelson@digium.com>
* pbx/pbx_spool.c: Convert pbx_spool to use string fields instead
of statically-sized buffers. In tests run after making this
conversion, I noticed an approximate 85% reduction in memory
usage for call file processing. Review:
http://reviewboard.digium.com/r/168/
2009-03-03 16:47 +0000 [r179742] Russell Bryant <russell@digium.com>
* main/channel.c, /: Merged revisions 179741 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r179741 | russell | 2009-03-03 10:45:46 -0600 (Tue, 03 Mar 2009)
| 6 lines Ensure chan->fdno always gets reset to -1 after
handling a channel fd event. Since setting fdno to -1 had to be
moved, a couple of other code paths that do process an fd event
return early and do not pass through the code path where it was
moved to. So, set it to -1 in a few other places, too. ........
2009-03-03 15:13 +0000 [r179675] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Please prefix default values with DEFAULT
2009-03-03 14:40 +0000 [r179672] Joshua Colp <jcolp@digium.com>
* main/channel.c, /: Merged revisions 179671 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r179671 | file | 2009-03-03 10:38:09 -0400 (Tue, 03 Mar 2009) | 3
lines Move where fdno is set to the default value to *after* the
read callback of the channel driver is called. We have to do this
as the underlying channel driver may need the fdno value to
determine what to read. ........
2009-03-03 13:54 +0000 [r179609] Russell Bryant <russell@digium.com>
* main/channel.c, /: Merged revisions 179608 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r179608 | russell | 2009-03-03 07:53:52 -0600 (Tue, 03 Mar 2009)
| 9 lines Make it easier to detect an improper call to
ast_read(). When you call ast_waitfor() on a channel, the index
into the channel fds array that holds the file descriptor that
poll() determines has input available is stored in fdno. This
patch clears out this value after a call to ast_read() and also
reports errors if ast_read() is called without an fdno set. From
a discussion on the asterisk-dev list. ........
2009-03-03 00:01 +0000 [r179537] Jeff Peeler <jpeeler@digium.com>
* main/channel.c, /: Merged revisions 179536 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r179536 | jpeeler | 2009-03-02 17:54:39 -0600 (Mon, 02 Mar 2009)
| 15 lines Fix bridging regression from commit 176701 This fixes
a bad regression where the bridge would exit after an attended
transfer was made. The problem was due to nexteventts getting set
after the masquerade which caused the bridge to return
AST_BRIDGE_COMPLETE. (closes issue #14315) Reported by:
tim_ringenbach ........
2009-03-02 23:36 +0000 [r179533] Russell Bryant <russell@digium.com>
* /, apps/app_meetme.c: Merged revisions 179532 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r179532 | russell | 2009-03-02 17:34:13 -0600 (Mon, 02 Mar 2009)
| 40 lines Move ast_waitfor() down to avoid the results of the
API call becoming stale. This call to ast_waitfor() was being
done way too soon in this section of code. Specifically, there
was code in between the call to waitfor and the code that uses
the result that puts the channel in autoservice. By putting the
channel in autoservice, the previous results of ast_waitfor()
become meaningless, as the autoservice thread will do it's own
ast_waitfor() and ast_read() on the channel. So, when we came
back out of autoservice and eventually hit the block of code that
calls ast_read() on the channel, there may not actually be any
input on the channel available. Even though the previous call to
ast_waitfor() in app_meetme said there was input, the autoservice
thread has since serviced the channel for some period of time.
This bug manifested itself while dvossel was doing some testing
of MeetMe in Asterisk trunk. He was using the timerfd timing
module. When the code hit ast_read() erroneously, it determined
that it must have been called because of input on the timer fd,
as chan->fdno was set to AST_TIMING_FD, since that was the cause
of the last legitimate call to ast_read() done by autoservice. In
this test, an IAX2 channel was calling into the MeetMe
conference. It was _much_ more likely to be seen with an IAX2
channel because of the way audio is handled. Every audio frame
that comes in results in a call to ast_queue_frame(), which then
uses ast_timer_enable_continuous() to notify the channel thread
that a frame is waiting to be handled. So, the chances of
ast_waitfor() indicating that a channel needs servicing due to a
timer event on an IAX2 event is very high. Finally, it is
interesting to note that if a different timing interface was
being used, this bug would probably not be noticed. When
ast_read() is called and erroneously thinks that there is a timer
event to handle, it calls the ast_timer_ack() function. The
pthread and dahdi timing modules handle the ack() function being
called when there is no event by simply ignoring it. In the case
of the timerfd module, it results in a read() on the timer fd
that will block forever, as there is no data to read. This caused
Asterisk to lock up very quickly. Thanks to dvossel and
mmichelson for the fun debugging session. :-) ........
2009-03-02 23:10 +0000 [r179469] Tilghman Lesher <tlesher@digium.com>
* /, main/app.c: Merged revisions 179468 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r179468 | tilghman | 2009-03-02 17:09:01 -0600 (Mon, 02 Mar 2009)
| 10 lines When ending a recording with silence detection,
remember to reduce the duration. The end of the recording is
correspondingly trimmed, but the duration was not trimmed by the
number of seconds trimmed, so the saved duration was necessarily
longer than the actual soundfile duration. (closes issue #14406)
Reported by: sasargen Patches: 20090226__bug14406.diff.txt
uploaded by tilghman (license 14) Tested by: sasargen ........
2009-03-02 23:06 +0000 [r179462-179465] Russell Bryant <russell@digium.com>
* res/res_timing_timerfd.c: Fix a reference leak in
timerfd_set_rate(). (found during a debugging session with
dvossel and mmichelson.)
* main/channel.c, /: Merged revisions 179461 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r179461 | russell | 2009-03-02 16:58:18 -0600 (Mon, 02 Mar 2009)
| 8 lines Ensure that only one thread is calling ast_settimeout()
on a channel at a time. For example, with an IAX2 channel, you
can have both the channel thread and the chan_iax2 processing
threads calling this function, and doing so twice at the same
time is a bad thing. (Found in a debugging session with dvossel
and mmichelson) ........
2009-03-02 20:16 +0000 [r179396] Jason Parker <jparker@digium.com>
* /, main/editline/configure, main/editline/np/unvis.c,
main/editline/sys.h, main/editline/configure.in: Merged revisions
179395 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r179395 | qwell | 2009-03-02 14:14:57 -0600 (Mon, 02 Mar 2009) |
1 line Remove several silly warnings in editline. One about a
broken preprocessor directive, and another about strlcpy/strlcat.
(closes issue #14264) Reported by: dimas ........
2009-03-02 17:18 +0000 [r179361] Tilghman Lesher <tlesher@digium.com>
* cdr/cdr_sqlite3_custom.c: Backport 1.6.0 fix to trunk (failsafe
if db is not loaded)
2009-03-02 14:28 +0000 [r179291-179323] Joshua Colp <jcolp@digium.com>
* channels/chan_iax2.c: Do not try to remove a registration
scheduled item if the scheduler context has already been
destroyed. (closes issue #14580) Reported by: alecdavis
* main/audiohook.c: Fix issue where changing the volume of both
directions of audio did not work. (closes issue #14574) Reported
by: KNK Patches: audiohook_volume_fix.diff uploaded by KNK
(license 545)
2009-03-01 23:25 +0000 [r179219-179254] Mark Michelson <mmichelson@digium.com>
* apps/app_speech_utils.c: Swap reversed timevals. This was pointed
out by ScribbleJ in #asterisk-dev. Thanks very much, ScribbleJ!
* channels/chan_sip.c: Properly free memory and remove scheduler
entries when a transmission failure occurs. Previously, only the
"data" field of the sip_pkt created during __sip_reliable_xmit
was freed when XMIT_ERROR was returned by __sip_xmit. When
retrans_pkt was called, this inevitably resulted in the reading
and writing of freed memory. XMIT_ERROR is a condition meaning
that we don't want to attempt resending the packet at all. The
proper action to take is to remove the scheduler entry we just
created, free the packet's data as well as the packet itself, and
unlink it from the list of packets on the sip_pvt structure.
(closes issue #14455) Reported by: Nick_Lewis Patches:
14455.patch uploaded by mmichelson (license 60) Tested by:
Nick_Lewis
2009-02-27 21:47 +0000 [r179164] Russell Bryant <russell@digium.com>
* res/res_ais.c, doc/distributed_devstate.txt,
configs/ais.conf.sample: Mark res_ais as experimental, as the
binary event format is subject to change.
2009-02-27 21:32 +0000 [r179161] Tilghman Lesher <tlesher@digium.com>
* cdr/cdr_sqlite3_custom.c: If config file is blank, don't load
module. (Closes issue #14563)
2009-02-27 21:23 +0000 [r179154] Russell Bryant <russell@digium.com>
* UPGRADE.txt: Add a note about the ordering of entries in sip.conf
in 1.6.1.
2009-02-27 20:34 +0000 [r179122] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_skinny.c: Add reload support to chan_skinny.
Special thanks goes to DEA who had to redo this patch twice
because we first put unload/load support in and later redid the
way we configure devices and lines. (closes issue #10297)
Reported by: DEA Patches: skinny-reload-trunkv2.diff uploaded by
wedhorn (license 30) skinny-reload-trunk-v4.txt uploaded by DEA
(license 3) With mods by me based on feedback from wedhorn and
Russell and seanbright Tested by: DEA, mvanbaak, pj Review:
http://reviewboard.digium.com/r/130/
2009-02-27 19:04 +0000 [r179057] Jason Parker <jparker@digium.com>
* doc/tex/channelvariables.tex: Update documentation for DIALEDTIME
and ANSWEREDTIME variables. (closes issue #14566) Reported by:
klaus3000 Patches: ANSWEREDTIME-1.4-patch.txt uploaded by
klaus3000 (license 65) ANSWEREDTIME-trunk-patch.txt uploaded by
klaus3000 (license 65)
2009-02-27 15:51 +0000 [r179021] Russell Bryant <russell@digium.com>
* sounds/Makefile: Fix downloading SIREN7 and SIREN14 sound
packages. In passing, also fix downloading SLIN16 extra sound
packages. (closes issue #14565) Reported by: jtodd
2009-02-27 03:45 +0000 [r178986] Steve Murphy <murf@digium.com>
* /, main/features.c, configs/features.conf.sample: Merged
revisions 178956 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 In this
case, it's just a matter of reducing the default timeouts from
2000 to 1000 msec, as the max def feature digit timeout is no
longer halved. ........ r178956 | murf | 2009-02-26 14:27:32
-0700 (Thu, 26 Feb 2009) | 18 lines This change moves the default
feature digit timeout to 1000 ms from the previous default of
500. As per bug 14515, a dev discussion arrived at a "mediated
concensus" of a default feature digit timeout of 1.0 sec. Some
voted for 1300; ctooley thought 1500 for distracted phone users
in phone booths; kpfleming put his foot down at 1.0 sec. Users
who found the previous default max delay of 250 msec perfect, are
welcome to override the new default. Notice that I said that 250
msec was the default; wait a minute, you might say, the config
file said it was 500 msec!; well, because of the bug fix for
14515, we found that 500 msec was actually enforcing a max of
250. The bug fix would restore 500 msec, but we felt even that
was a bit tight for most users... 2000 msec was pushed earlier by
mmichelson, so that reduces to 1000 msec after the bug fix.
Enjoy! ........
2009-02-26 18:41 +0000 [r178919] Tilghman Lesher <tlesher@digium.com>
* main/features.c, CHANGES, configs/features.conf.sample: Sound
confirmation of call pickup success. (closes issue #13826)
Reported by: azielke Patches: pickupsound2-trunk.patch uploaded
by azielke (license 548) __20081124_bug_13826_updated.patch
uploaded by lmadsen (license 10) Tested by: lmadsen
2009-02-26 17:46 +0000 [r178871] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c: IAX2 prune realtime, minor tweak to last
fix A return statement was missing which caused unexpected cli
output. issue #14479
2009-02-26 17:45 +0000 [r178828-178870] Steve Murphy <murf@digium.com>
* apps/app_osplookup.c, apps/app_rpt.c: These small fixes prevent
compiler warnings with ubuntu 8.10's gcc-4.3.2, which tend to
break my dev-mode build. Not a problem in 1.6.x.
* /, main/features.c: Merged revisions 178804 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r178804 | murf | 2009-02-26 10:09:03 -0700 (Thu, 26 Feb 2009) |
28 lines This patch prevents the feature detection timeout from
being cut in half. Because the ast_channel_bridge() call will
return 0 and pass a frame pointer for both DTMF_BEGIN and
DTMF_END, the feature_timer field in hte config struct is getting
decremented twice, which effectively cuts the digittimeout in
half. I added conditions to the if statement to only let DTMF_END
frames to flow thru, which solved the problem. Also, when the
frame pointer is null, let control flow thru-- this usually
happens on timeouts. I added a comment to the code to explain
what's going on and why. Many thanks to sodom for reporting this
problem. Personnally, it always seemed like something was wrong
with the featuredigittimeout, but I never could quite decide
what... and was too busy to investigate. This bug forced the
issue, and now we know. Sodom had other issues in 14515, but I
couldn't reproduce them. If he still has problems, and wants to
get them solved, he is welcome to reopen 14515. (closes issue
#14515) Reported by: sodom Patches: 14515.patch uploaded by murf
(license 17) Tested by: murf, sodom ........
2009-02-26 16:42 +0000 [r178801] Joshua Colp <jcolp@digium.com>
* main/file.c: Fix an issue where the timer for file playback would
not be stopped if DAHDI was not installed. (closes issue #14541)
Reported by: grant
2009-02-26 15:50 +0000 [r178767] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c: IAX2 prune realtime fix Iax2 prune realtime
had issues. If "iax2 prune realtime all" was called, it would
appear like the command was successful, but in reality nothing
happened. This is because the reload that was supposed to take
place checks the config files, sees no changes, and does nothing.
If there had been a change in the the config file, the realtime
users would have been marked for deletion and everything would
have been fine. Now prune_users() and prune_peers() are called
instead of reload_config() to prune all users/peers that are
realtime. These functions remove all users/peers with the
rtfriend and delme flags set. iax2_prune_realtime() also lacked
the code to properly delete a single friend. For example. if iax2
prune realtime <friend> was called, only the peer instance would
be removed. The user would still remain. (closes issue #14479)
Reported by: mousepad99 Review:
http://reviewboard.digium.com/r/176/
2009-02-26 15:40 +0000 [r178764] Joshua Colp <jcolp@digium.com>
* main/indications.c: Ensure there is a valid tone part before
trying to play tones. (closes issue #14558) Reported by:
alecdavis
2009-02-26 15:02 +0000 [r178733] Olle Johansson <oej@edvina.net>
* configs/res_snmp.conf.sample: Clarifications on the different
models and reference to further docs.
2009-02-26 13:39 +0000 [r178703-178704] Kevin P. Fleming <kpfleming@digium.com>
* README: another minor commit to test post-commit script changes
(now testing post-revprop-change as well, third try)
* README: minor commit to test post-commit script changes
2009-02-25 19:49 +0000 [r178573-178607] Tilghman Lesher <tlesher@digium.com>
* main/stdtime/localtime.c: Picky, picky buildbots
* configure, include/asterisk/autoconfig.h.in, configure.ac,
main/stdtime/localtime.c: Use notification when timezone files
change and re-scan then. (closes issue #14300) Reported by:
jamessan Patches: 20090127__bug14300.diff.txt uploaded by
tilghman (license 14) 20090224__bug14300.diff uploaded by
jamessan (license 246) Tested by: jamessan Review:
http://reviewboard.digium.com/r/136/
* res/res_odbc.c: Oops, wrong direction of command
2009-02-25 12:45 +0000 [r178509] Russell Bryant <russell@digium.com>
* /, main/asterisk.c: Merged revisions 178508 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r178508 | russell | 2009-02-25 06:43:36 -0600 (Wed, 25 Feb 2009)
| 2 lines Update the copyright year for the main page of the
doxygen documentation. ........
2009-02-24 23:27 +0000 [r178375-178446] Tilghman Lesher <tlesher@digium.com>
* /, configs/extensions.conf.sample: Merged revisions 178445 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r178445 | tilghman | 2009-02-24 17:25:24 -0600 (Tue, 24 Feb 2009)
| 5 lines Add section about the #exec command in configuration
files. (closes issue #14540) Reported by: jtodd Patch by: jtodd,
with additional notes by tilghman (license 14) ........
* main/asterisk.c: Apparently, a void cast doesn't override
warn_unused_result.
* main/asterisk.c: The 3 possible errors with pipe(2) are all
impossible in this situation.
2009-02-24 20:39 +0000 [r178374] Russell Bryant <russell@digium.com>
* /, main/rtp.c: Merged revisions 178373 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r178373 | russell | 2009-02-24 14:36:19 -0600 (Tue, 24 Feb 2009)
| 6 lines Only set dtmfcount on BEGIN, and ensure it gets reset
to 0 properly. (issue #14460) Reported by: moliveras Tested by:
russell ........
2009-02-24 20:06 +0000 [r178303-178342] Tilghman Lesher <tlesher@digium.com>
* utils/astcanary.c, main/asterisk.c: Use a SIGPIPE to kill the
process, instead of depending upon the astcanary process being
inherited by init.
* utils/astcanary.c: Cause astcanary to exit if Asterisk exits
abnormally and doesn't kill astcanary. Also, add some
documentation supporting the use of astcanary. (closes issue
#14538) Reported by: KNK Patches: asterisk-1.6.x-astcanary.diff
uploaded by KNK (license 545)
2009-02-24 17:42 +0000 [r178300] David Vossel <dvossel@digium.com>
* doc/manager_1_1.txt, CHANGES, channels/chan_iax2.c: Allows
manager command to see if IAX link is trunked and encrypted.
Displays what kind of encryption is enabled as well. Manager
command "iaxpeers" now shows if a link is trunked and encrypted.
Instead of encryption saying simply "yes" or "no", it now
displays what type of encryption is enabled and if keyrotation is
on or not. (closes issue #14427) Reported by: snuffy Patches:
iax_show_trunks.diff uploaded by snuffy (license 35)
2009022200_iax2_show_trunkencryption.diff.txt uploaded by
mvanbaak (license 7) Tested by: mvanbaak, dvossel, snuffy Review:
http://reviewboard.digium.com/r/173/
2009-02-24 15:18 +0000 [r178213] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 178205 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r178205 | file | 2009-02-24 11:16:07 -0400 (Tue, 24 Feb 2009) | 9
lines Skip check for extension when subscribing for MWI. Since
the remote side is not actually subscribing to a specific
extension when subscribing for MWI just skip the check to see if
the extension exists. They can't use it to specify the mailbox
either since we require configuration of that in sip.conf (closes
issue #14531) Reported by: festr ........
2009-02-23 23:11 +0000 [r178142] Russell Bryant <russell@digium.com>
* /, main/rtp.c: Merged revisions 178141 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r178141 | russell | 2009-02-23 17:09:01 -0600 (Mon, 23 Feb 2009)
| 14 lines Fix infinite DTMF when a BEGIN is received without an
END. This commit is related to rev 175124 of 1.4 where a previous
attempt was made to fix this problem. The problem with the
previous patch was that the inserted code needed to go _before_
setting the lastrxts to the current timestamp. Because those were
the same, the dtmfcount variable was never decremented, and so
the END was never sent. In passing, I removed the dtmfsamples
variable which was completed unused. I also removed a redundant
setting of the lastrxts variable. (closes issue #14460) Reported
by: moliveras ........
2009-02-23 21:02 +0000 [r178107] Tilghman Lesher <tlesher@digium.com>
* configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c:
Permit emailsubject and emailbody to be set per mailbox. (closes
issue #14372) Reported by: fhackenberger Patches:
voicemail_individual_subject_and_body_1.6.1 uploaded by
fhackenberger (license 592) with additional fixes by Corydon76
(license 14)
2009-02-23 18:23 +0000 [r178061] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_skinny.c: update the new manager commands in
chan_skinny to match chan_sip's headers. requested by oej.
2009-02-23 17:59 +0000 [r178030] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c: Changes the way keyrotation is enabled by
default Key rotation was enabled by default by setting the global
encryption method to IAX_ENCRYPT_KEYROTATE. the problem with this
is that if encryption is not enabled, and the encryption method
is set to anything except 0, the peer appears to have encryption
enabled when issuing a "iax2 show peers". Rather than have the
key rotation bit always set by default, it is now only set when
an encryption method is enabled. (closes issue #14523) Reported
by: mvanbaak
2009-02-23 17:48 +0000 [r178027] Michiel van Baak <michiel@vanbaak.info>
* CHANGES: list the addition of the SKINNY manager actions in the
CHANGES file.
2009-02-23 17:29 +0000 [r178022] Russell Bryant <russell@digium.com>
* tests/test_sched.c, main/sched.c: Fix a regression in scheduler
entry ordering, and add a regression test for it. (closes issue
#14522) Reported by: pj Tested by: russell
2009-02-22 23:04 +0000 [r177988] Michiel van Baak <michiel@vanbaak.info>
* doc/manager_1_1.txt, channels/chan_skinny.c: Add a couple of
manager commands to chan_skinny Added: SKINNYdevices
SKINNYshowdevice SKINNYlines SKINNYshowline (closes issue #14521)
Reported by: mvanbaak Review:
http://reviewboard.digium.com/r/170/
2009-02-21 15:59 +0000 [r177944] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c: On update, test against the existence of
sipregs.
2009-02-21 14:37 +0000 [r177913] Michiel van Baak <michiel@vanbaak.info>
* main/asterisk.c: add extra check for sysinfo/sysctl (closes issue
#14513) Reported by: snuffy Patches: bug14513_fixsysinfo.diff
uploaded by snuffy (license 35)
2009-02-21 14:16 +0000 [r177884] Sean Bright <sean.bright@gmail.com>
* main/hashtab.c, include/asterisk/hashtab.h: Trailing whitespace,
minor coding guideline fixes, and start beefing up the hashtab
documentation a bit.
2009-02-21 13:17 +0000 [r177855] Russell Bryant <russell@digium.com>
* include/asterisk/indications.h: Fix build issues on Solaris and
OpenBSD. (closes issue #14512) Reported by: snuffy
2009-02-21 13:13 +0000 [r177849-177852] Michiel van Baak <michiel@vanbaak.info>
* Makefile, contrib/init.d/rc.debian.asterisk,
contrib/init.d/rc.archlinux.asterisk,
contrib/scripts/safe_asterisk: set
ASTVARRUNDIR=$(localstatedir)/run/asterisk as default path When
running asterisk as non-root and without this patch the pidfile
wants to go into /var/run/asterisk.pid. This directory is not
writable for the non-root user and changing permissions is not an
option. Putting it in /var/run/asterisk/asterisk.pid makes it
possible to set permissions on the /var/run/asterisk dir so
everything works as it should be. Patched committed is based on
pabelanger's patch. (closes issue #13153) Reported by: pabelanger
Patches: 2009012900_bug13153-nonrootscripts.diff.txt uploaded by
mvanbaak (license 7) Review: http://reviewboard.digium.com/r/139/
* channels/chan_sip.c: make chan_sip.c compile on OpenBSD again.
2009-02-20 23:02 +0000 [r177732-177787] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c, /: Merged revisions 177786 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r177786 | tilghman | 2009-02-20 16:59:52 -0600 (Fri, 20 Feb 2009)
| 9 lines Don't print the CR-NL combination when we aren't
outputting to the manager. An embedded CR-NL in a CLI command
screws up several AMI parsers that don't expect to see that
combination in the middle of output. (Closes issue #14305)
Reported by: martins Patch by: tilghman ........
* /, include/asterisk/threadstorage.h: Merged revisions 177701 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r177701 | tilghman | 2009-02-20 15:15:01 -0600 (Fri, 20 Feb 2009)
| 3 lines This exception does not appear to still be true for
Solaris 10, and OpenSolaris definitely needs it to be removed.
Fixed for snuff-home on -dev channel. ........
2009-02-20 20:29 +0000 [r177699] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
* apps/app_fax.c: Make app_fax compatible with spandsp-0.0.6pre4
Prior to spandsp-0.0.6pre4 the t30_stats_t structure used a
pages_transferred integer to indicate the number of pages
transferred (so far) during the fax session. The
spandsp-0.0.6pre4 release removed the pages_transferred integer
and replaced it with two different integers - pages_tx and
pages_rx. This revision uses the new integers for
spandsp-0.0.6pre4 while maintaining backwards compatibility for
previous spandsp releases.
2009-02-20 17:29 +0000 [r177661-177664] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/app.h, main/app.c, apps/app_system.c: Allow
semicolons to be escaped, when passing arguments to the System
command. (closes issue #14231) Reported by: jcovert Patches:
20090113__bug14231__2.diff.txt uploaded by Corydon76 (license 14)
corrected_20090113__bug14231__2.diff.txt uploaded by jcovert
(license 551) Tested by: jcovert
* apps/app_voicemail.c: Oops, merge broke trunk
2009-02-20 00:35 +0000 [r177624] Jeff Peeler <jpeeler@digium.com>
* channels/chan_sip.c: Set sip_request ast_str data to NULL so
ast_str_copy allocates space properly in copy_request (issue
#14478) Reported by: erik_dedecker
2009-02-19 23:56 +0000 [r177595] Steve Murphy <murf@digium.com>
* /, main/Makefile, main/ast_expr2f.c: Merged revisions 177540 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 Trunk was
already pretty 8-bit clean; but I'm still removing the --full
from the flex command so everything is uniform. ........ r177540
| murf | 2009-02-19 15:51:37 -0700 (Thu, 19 Feb 2009) | 21 lines
This patch fixes a problem with 8-bit input to the ast_expr2
scanner. The real culprit was the --full argument to flex in the
Makefile! This causes a 7-bit scanner to be generated. I reviewed
the rules and found one rule where I needed to specifically
include 8-bit chars for a token. I tested against the text
supplied by ibercom, and all looks very well. This has been there
a surprisingly long time! (closes issue #14498) Reported by:
ibercom Patches: 14498.patch uploaded by murf (license 17) Tested
by: murf ........
2009-02-19 22:33 +0000 [r177506-177537] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_voicemail.c: Merged revisions 177536 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r177536 | tilghman | 2009-02-19 16:26:01 -0600 (Thu, 19
Feb 2009) | 7 lines Fix up potential crashes, by reducing the
sharing between interactive and non-interactive threads. (closes
issue #14253) Reported by: Skavin Patches:
20090219__bug14253.diff.txt uploaded by Corydon76 (license 14)
Tested by: Skavin ........
* doc/database_transactions.txt (added): Document how to use
database transactions
2009-02-19 16:45 +0000 [r177387] Jeff Peeler <jpeeler@digium.com>
* include/asterisk/channel.h: Fix another merge error from 176708
2009-02-19 16:38 +0000 [r177384] Joshua Colp <jcolp@digium.com>
* /, apps/app_speech_utils.c: Merged revisions 177383 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r177383 | file | 2009-02-19 12:37:25 -0400 (Thu, 19 Feb
2009) | 3 lines If we are able to create a speech structure unset
the ERROR variable in case it was previously set. (issue
#LUMENVOX-13) ........
2009-02-19 15:56 +0000 [r177356] Jeff Peeler <jpeeler@digium.com>
* main/features.c: Fix mismerge from revision 176708 pointed out by
Kaloyan Kovachev on the asterisk-dev mailing list. Thanks!
2009-02-19 00:26 +0000 [r177320] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/res_odbc.h, funcs/func_odbc.c, CHANGES,
res/res_odbc.c, configs/res_odbc.conf.sample: ODBC transaction
support
2009-02-19 00:08 +0000 [r177291] Joshua Colp <jcolp@digium.com>
* CHANGES: Update CHANGES file to include MWI subscription support
that was added some time ago.
2009-02-18 23:51 +0000 [r177287] Tilghman Lesher <tlesher@digium.com>
* main/strings.c: Handle negative length and eliminate a condition
that is always true.
2009-02-18 23:50 +0000 [r177286] Steve Murphy <murf@digium.com>
* /, res/ael/ael.tab.c, res/ael/ael.y: Merged revisions 177225 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r177225 | murf | 2009-02-18 15:43:14 -0700 (Wed, 18 Feb 2009) |
34 lines This patch fixes a regression of sorts that was
introduced in rev 24425. It basically fixes AST-190/ABE-1782.
What was wrong: the user has 6000 extensions in one context; and
then 6000 contexts, one per extension. The parser could only
handle about 4893 of the 6000 extens in the single context. This
was due to the regression I mentioned. To get rid of shift/reduce
conflicts, Luigi set up right-recursive lists for globals,
context elements, switch lists, and statements. Right recursive
lists got rid of the warnings, but instead, they use up a
tremendous amount of stack space when the lists are long. I saw
this a few years back, and resolved not to fix it until someone
complained. That day has arrived! After the changes were made, I
ran the regression test suite, and there were no problems. I took
the test case the user provided, and added 100,000 extensions to
the single context, that already had 6,000 extens in it. (I'll
see your 6, and raise you 100!) It takes a few minutes to read it
all in, check it and generate code for it, but no problems. So, I
think I can say that fundamentally, there are no longer any
limits on the number of items you can place in contexts,
statement blocks, switches, or globals, beyond your virt mem
constraints. ........
2009-02-18 23:09 +0000 [r177229] Kevin P. Fleming <kpfleming@digium.com>
* main/frame.c: fix two very minor bugs: if anyone ever uses
SLINEAR16 as a format in RTP, ensure that the samples are
byte-swapped to network order if needed. also, when a smoother is
operating on a format that has a sample rate other than 8000
samples per second, use the proper sample rate for computing
delivery timestamps.
2009-02-18 22:51 +0000 [r177226] David Vossel <dvossel@digium.com>
* main/features.c: Locking issue in action_bridge and bridge_exec
action_bridge() and bridge_exec() both search for the channels to
bridge to, and then immediately drop the lock. Instead, they
should hold the lock until the masquerade is complete. This will
guarantee the channel remains and prevent any other weirdness
from occurring. In action_bridge() some more weirdness comes into
play. Both channels are needlessly locked at the same time and
perform the exact same logic. It makes sense from a coding
organizational standpoint, but could cause a theoretical deadlock
so I split the code up. There is an issue associated with this,
but since its a rather complicated thing to reproduce I'm not
certain this alone will close it. issue# 14296 Review:
http://reviewboard.digium.com/r/167/
2009-02-18 20:11 +0000 [r177162] Jeff Peeler <jpeeler@digium.com>
* channels/h323/compat_h323.cxx, autoconf/ast_check_pwlib.m4,
channels/h323/cisco-h225.h, channels/h323/caps_h323.cxx,
channels/h323/ast_h323.cxx, channels/h323/ast_ptlib.h (added),
configure, channels/h323/compat_h323.h, configure.ac,
channels/h323/caps_h323.h, autoconf/ast_prog_sed.m4,
channels/h323/ast_h323.h, channels/h323/chan_h323.h,
channels/h323/cisco-h225.cxx: Modify h323 to build against PTLib
as well as the older PWLib Several changes in PTLib have occurred
requiring build time detection. Changes accounted for include the
library name change, config option change, install location
change, and a boolean type change which is handled by
ast_ptlib.h. Also, the sed check has been modified to properly
work with autoconf >= 2.62. (closes issue #14224) Reported by:
bergolth Patches: asterisk-autoconf-sed.patch uploaded by
bergolth (license 661) asterisk-pwlib-v3.patch uploaded by
bergolth (license 661) Tested by: jpeeler
2009-02-18 19:12 +0000 [r177101] Russell Bryant <russell@digium.com>
* apps/app_meetme.c: Re-add 'o' option to MeetMe, reverting rev
62297. Enabling this option by default proved to be a bad idea,
as the talker detection is not very reliable. So, make it
optional again, and off by default. (issue #13801) Reported by:
justdave
2009-02-18 19:05 +0000 [r177098] Tilghman Lesher <tlesher@digium.com>
* /, include/asterisk/config.h: Merged revisions 177096 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r177096 | tilghman | 2009-02-18 12:30:38 -0600 (Wed, 18 Feb 2009)
| 2 lines Document the return value of the update method (as
requested on -dev list) ........
2009-02-18 17:24 +0000 [r177035] Doug Bailey <dbailey@digium.com>
* main/utils.c: Fixed error where a check for an zero length,
terminated string was needed.
2009-02-18 17:11 +0000 [r177005] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Fix ordering of output for a ChannelUpdate
manager event. (closes issue #14497) Reported by: vinsik Patches:
chan_update_fix-chan_sip.c.diff uploaded by vinsik (license 623)
2009-02-18 16:09 +0000 [r176948] Doug Bailey <dbailey@digium.com>
* main/utils.c: Need to take into account the \0 terminator of the
old string to determine the amount available.
2009-02-18 15:35 +0000 [r176943] Steve Murphy <murf@digium.com>
* main/pbx.c: This patch fixes merge_contexts_and_delete so it does
not deadlock when hints are present. Reason: when I re-engineered
the merge_and_delete func to reduce its lock time, I failed to
notice that the functions it calls still also do locking as
before. This leads to deadlocks on dialplan reloads, when there
are actually living, subscribed hints registered in the system.
While the reporter come across this problem while using AEL, I
might note that these deadlocks should also happen if
extensions.conf were used. Here I added these routines to pbx.c:
ast_add_extension_nolock add_pri_lockopt
ast_add_extension2_lockopt find_context add_hint_nolock All of
the above routines are static and restricted to be used only
within pbx.c, and more specifically within the
merge_contexts_and_delete routine. They are pretty much the same
as their counterparts except they don't lock contexts or hints.
Most of them now do the real work of their name-alike, with
optional locking via extra arguments, and are called by their
name-alike. The goal was to have the original functions so they
would behave exactly as before. Both PJ and I tested these fixes,
and the deadlocking problem is no longer encountered. (closes
issue #14357) Reported by: pj Patches: 14357.diff uploaded by
murf (license 17) Tested by: pj, murf
2009-02-18 06:14 +0000 [r176901-176904] Russell Bryant <russell@digium.com>
* include/asterisk/heap.h: Add example code for a heap traversal.
* main/pbx.c: Fix a number of incorrect uses of strncpy(). The big
problem here is that the 3rd argument provided in these uses of
strncpy() did not reserve a byte for the null terminator, leaving
the potential for writing one byte past the end of the buffer.
Aside from this, there were coding guidelines violations with
regards to spacing, as well as hard coded lengths being used
instead of sizeof().
2009-02-18 02:55 +0000 [r176869] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
* channels/chan_sip.c: T38 faxdetect should jump to the 'fax'
extension for incoming calls only The previous implementation of
T38 faxdetect resulted in both sides of the call jumping to a fax
extension when both sides had 't38pt_udptl=yes' and
'faxdetect=yes' in sip.conf and a 'fax' extension in the current
context. This revision will jump to a 'fax' extension on incoming
calls only.
2009-02-18 02:02 +0000 [r176841] Kevin P. Fleming <kpfleming@digium.com>
* main/rtp.c: suppress smoothers for Siren codecs as well as Speex
and G.723.1
2009-02-17 22:52 +0000 [r176771] Russell Bryant <russell@digium.com>
* apps/app_milliwatt.c: Remove a dependency that no longer exists.
2009-02-17 22:28 +0000 [r176760] Shaun Ruffell <sruffell@digium.com>
* codecs/codec_dahdi.c: Several changes to codec_dahdi to play nice
with G723. This commit brings in the changes that were living out
on the svn/asterisk/team/sruffell/asterisk-trunk-transcoder
branch. codec_dahdi.c now always uses signed linear as the simple
codec so that a soft g729 codec will not end up being preferred
to the hardware codec. There are also changes to allow
codec_dahdi.c to feed packets to the hardware in the native
sample size of the codec. This solves problems with choppy audio
when using G723.
2009-02-17 22:08 +0000 [r176708] Jeff Peeler <jpeeler@digium.com>
* main/channel.c, /, main/features.c, include/asterisk/channel.h:
Merged revisions 176701 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r176701 | jpeeler | 2009-02-17 15:54:34 -0600 (Tue, 17 Feb 2009)
| 17 lines Modify bridging to properly evaluate DTMF after first
warning is played The main problem is currently if the Dial flag
L is used with a warning sound, DTMF is not evaluated after the
first warning sound. To fix this, a flag has been added in
ast_generic_bridge for playing the warning which ensures that if
a scheduled warning is missed, multiple warrnings are not played
back (due to a feature evaluation or waiting for digits).
ast_channel_bridge was modified to store the nexteventts in the
ast_bridge_config structure as that information was lost every
time ast_channel_bridge was reentered, causing a hangup due to
incorrect time calculations. (closes issue #14315) Reported by:
tim_ringenbach Reviewed on reviewboard:
http://reviewboard.digium.com/r/163/ ........
2009-02-17 22:02 +0000 [r176706] Mark Michelson <mmichelson@digium.com>
* tests/test_sched.c: Use constants from inttypes.h to clear up
32-bit compilation errors
2009-02-17 21:59 +0000 [r176705] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
* channels/chan_sip.c: create a UDPTL structure in
create_addr_from_peer() if it does not already exist for T38 This
is required to create a UDPTL structure in
create_addr_from_peer() to handle the scenario where
't38pt_udptl=yes' is not defined in the [general] section of
sip.conf but is defined the peer's context. I tested this patch
by enabling t38pt_udptl in the [general] section on one system
and only enabling t38pt_udptl in a peer's context on the system
sending a fax. Without the patch, the sending system will fail to
initiate T38 negotiation with the warning message, "No way to add
SDP without an UDPTL structure". When this patch is applied the
sending side will successfully initiate T38 negotiation.
2009-02-17 21:40 +0000 [r176697] Mark Michelson <mmichelson@digium.com>
* include/asterisk/frame.h: Clear up documentation of
AST_FRIENDLY_OFFSET in frame.h
2009-02-17 21:23 +0000 [r176669] Tilghman Lesher <tlesher@digium.com>
* /: Recorded merge of revisions 176661 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r176661 | tilghman | 2009-02-17 15:21:41 -0600 (Tue, 17 Feb 2009)
| 9 lines Backport change to 1.4: Prior to masquerade, move the
group definitions to the channel performing the masq, so that the
group count lingers past the bridge. (closes issue #14275)
Reported by: kowalma Patches: 20090216__bug14275.diff.txt
uploaded by Corydon76 (license 14) Tested by: kowalma ........
2009-02-17 21:22 +0000 [r176666] Russell Bryant <russell@digium.com>
* main/channel.c, res/res_timing_pthread.c, res/res_timing_dahdi.c,
res/res_timing_timerfd.c, include/asterisk/timing.h,
main/timing.c: Update the timing API to have better support for
multiple timing interfaces. 1) Add module use count handling so
that timing modules can be unloaded. 2) Implement unload_module()
functions for the timing interface modules. 3) Allow multiple
timing modules to be loaded, and use the one with the highest
priority value. 4) Report which timing module is being use in the
"timing test" CLI command. (closes issue #14489) Reported by:
russell Review: http://reviewboard.digium.com/r/162/
2009-02-17 21:14 +0000 [r176642] Tilghman Lesher <tlesher@digium.com>
* channels/chan_local.c: Prior to masquerade, move the group
definitions to the channel performing the masq, so that the group
count lingers past the bridge. (closes issue #14275) Reported by:
kowalma Patches: 20090216__bug14275.diff.txt uploaded by
Corydon76 (license 14) Tested by: kowalma
2009-02-17 21:04 +0000 [r176632-176639] Russell Bryant <russell@digium.com>
* tests/test_sched.c (added), main/sched.c: Significantly improve
scheduler performance under high load. This patch changes the
scheduler to use a max-heap to store pending scheduler entries
instead of a fully sorted doubly linked list. When the number of
entries in the scheduler gets large, this will perform much
better. For much more detailed information on this change, see
the review request. Review: http://reviewboard.digium.com/r/160/
* tests/test_heap.c (added): Add a test module for the heap
implementation. Review: http://reviewboard.digium.com/r/160/
* main/Makefile, main/heap.c (added), include/asterisk/heap.h
(added): Add an implementation of the heap data structure. A heap
is a convenient data structure for implementing a priority queue.
Code from svn/asterisk/team/russell/heap/. Review:
http://reviewboard.digium.com/r/160/
2009-02-17 20:50 +0000 [r176631] Olle Johansson <oej@edvina.net>
* include/asterisk/config.h: Typo
2009-02-17 20:41 +0000 [r176627] Russell Bryant <russell@digium.com>
* channels/chan_unistim.c, main/pbx.c, apps/app_read.c,
configs/indications.conf.sample, apps/app_playtones.c (added),
include/asterisk/indications.h, apps/app_readexten.c,
apps/app_disa.c, UPGRADE.txt, include/asterisk/channel.h,
include/asterisk/_private.h, main/indications.c, main/loader.c,
main/channel.c, channels/chan_misdn.c, channels/chan_skinny.c,
funcs/func_channel.c, res/snmp/agent.c, main/app.c,
res/res_indications.c (removed), main/asterisk.c: Merge a large
set of updates to the Asterisk indications API. This patch
includes a number of changes to the indications API. The primary
motivation for this work was to improve stability. The object
management in this API was significantly flawed, and a number of
trivial situations could cause crashes. The changes included are:
1) Remove the module res_indications. This included the critical
functionality that actually loaded the indications configuration.
I have seen many people have Asterisk problems because they
accidentally did not have an indications.conf present and loaded.
Now, this code is in the core, and Asterisk will fail to start
without indications configuration. There was one part of
res_indications, the dialplan applications, which did belong in a
module, and have been moved to a new module, app_playtones. 2)
Object management has been significantly changed. Tone zones are
now managed using astobj2, and it is no longer possible to crash
Asterisk by issuing a reload that destroys tone zones while they
are in use. 3) The API documentation has been filled out. 4) The
API has been updated to follow our naming conventions. 5) Various
bits of code throughout the tree have been updated to account for
the API update. 6) Configuration parsing has been mostly
re-written. 7) "Code cleanup" The code is from
svn/asterisk/team/russell/indications/. Review:
http://reviewboard.digium.com/r/149/
2009-02-17 18:49 +0000 [r176592] Tilghman Lesher <tlesher@digium.com>
* funcs/func_odbc.c, res/res_odbc.c: Add assertions in the quest to
track down a refcount leak. (closes issue #14485) Reported by:
davevg
2009-02-17 17:33 +0000 [r176557] Russell Bryant <russell@digium.com>
* main/pbx.c, apps/app_queue.c: Fix a race condition that caused
device states to become incorrect for hints. The problem here is
that the hint processing code was subscribed to the wrong event
type. So, it started processing state for a hint too soon, before
the device state cache had been updated. Also, fix a similar bug
in app_queue, as it was also subscribed to the wrong event type.
(closes issue #14461) Reported by: alecdavis
2009-02-17 17:28 +0000 [r176513-176556] Olle Johansson <oej@edvina.net>
* configs/extconfig.conf.sample: Typo
* main/config.c: If there are no realtime engines, there's no
reason to check for realtime families
2009-02-17 14:39 +0000 [r176360-176501] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c: In this version, we can combine the queries,
because we support dropping nonexistent columns.
* /, channels/chan_sip.c: Merged revisions 176426 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r176426 | tilghman | 2009-02-16 18:49:22 -0600 (Mon, 16 Feb 2009)
| 10 lines After a 'sip reload', qualifies for realtime peers
weren't immediately restarted, instead waiting until the next
registration. We're now caching the qualify across a
reload/restart and starting the qualify immediately upon loading
the peer. (closes issue #14196) Reported by: pdf Patches:
20090120__bug14196_1.4.diff.txt uploaded by pdf (license 663)
Tested by: pdf ........
* main/strings.c: Might want to update the buffer pointer after a
realloc (or we crash) (closes issue #14485) Reported by: davevg
2009-02-16 23:37 +0000 [r176356] Kevin P. Fleming <kpfleming@digium.com>
* sounds/sounds.xml: add support for Siren7 and Siren14 flavors of
prompts and music on hold
2009-02-16 23:33 +0000 [r176355] David Vossel <dvossel@digium.com>
* /, channels/chan_iax2.c: Merged revisions 176354 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r176354 | dvossel | 2009-02-16 17:30:52 -0600 (Mon, 16
Feb 2009) | 8 lines Fixes issue with AST_CONTROL_SRCUPDATE not
being relayed correctly during bridging This should have been
committed with rev176247, but I missed it. srcupdate frames no
longer break out of the native bridge, but are not being sent to
the other call leg either. This fixs that. issue #13749 ........
2009-02-16 23:14 +0000 [r176320] Tilghman Lesher <tlesher@digium.com>
* channels/chan_skinny.c: Use the correct list macros for deleting
an item from the middle of a list. (issue #13777) Reported by: pj
Patches: 20090203__bug13777.diff.txt uploaded by Corydon76
(license 14) Tested by: pj
2009-02-16 21:45 +0000 [r176255] Kevin P. Fleming <kpfleming@digium.com>
* /, main/utils.c, include/asterisk/stringfields.h: Merged
revisions 176216 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r176216 | kpfleming | 2009-02-16 15:10:38 -0600 (Mon, 16 Feb
2009) | 3 lines fix a flaw in the ast_string_field_build() family
of API calls; these functions made no attempt to reuse the space
already allocated to a field, so every time the field was written
it would allocate new space, leading to what appeared to be a
memory leak. ........ r176254 | kpfleming | 2009-02-16 15:41:46
-0600 (Mon, 16 Feb 2009) | 3 lines correct a logic error in the
last stringfields commit... don't mark additional space as
allocated if the string was built using already-allocated space
........
2009-02-16 21:40 +0000 [r176253] Mark Michelson <mmichelson@digium.com>
* /, apps/app_meetme.c: Merged revisions 176249,176252 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r176249 | mmichelson | 2009-02-16 15:34:27 -0600 (Mon,
16 Feb 2009) | 14 lines Open the DAHDI pseudo device and set it
to be nonblocking atomically Apparently on FreeBSD, attempting to
set the O_NONBLOCKING flag separately from opening the file was
causing an "inappropriate ioctl for device" error. While I cannot
fathom why this would be happening, I certainly am not opposed to
making the code a bit more compact/efficient if it also fixes a
bug. (closes issue #14482) Reported by: ys Patches: meetme.patch
uploaded by ys (license 281) Tested by: ys ........ r176252 |
mmichelson | 2009-02-16 15:39:21 -0600 (Mon, 16 Feb 2009) | 3
lines Remove unused variable and make dev-mode compilation happy
........
2009-02-16 21:30 +0000 [r176248] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c: Merged revisions 175597 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r175597 |
dvossel | 2009-02-13 14:11:55 -0600 (Fri, 13 Feb 2009) | 4 lines
Fixed iax2 key rotation backwards compatibility Turns key
rotation back on by default. Added bit into encryption IE to
indicate whether or not key rotation is supported or not. If it
is not supported then it is not enabled, which insures backwards
compatibility. This eliminates the need for the keyrotate option
in iax.conf, so it has been removed. ........
2009-02-16 18:25 +0000 [r176174] Mark Michelson <mmichelson@digium.com>
* main/logger.c: Assist proper thread synchronization when stopping
the logger thread. I was finding that on my dev box, occasionally
attempting to "stop now" in trunk would cause Asterisk to hang. I
traced this to the fact that the logger thread was waiting on a
condition which had already been signalled. The logger thread
also need to be sure to check the value of the
close_logger_thread variable. The close_logger_thread variable is
only checked when the list of logmessages is empty. This allows
for the logger thread to print and free any pending messages
before exiting.
2009-02-16 17:44 +0000 [r176138] Tilghman Lesher <tlesher@digium.com>
* channels/chan_dahdi.c: Can't set debug level 2 (intense
debugging) unless the syntax matches
2009-02-16 17:09 +0000 [r176100] Russell Bryant <russell@digium.com>
* channels/chan_features.c (removed): Remove chan_features. Review:
http://reviewboard.digium.com/r/161/
2009-02-16 15:36 +0000 [r176030] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 176029 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r176029 | file | 2009-02-16 11:33:53 -0400 (Mon, 16 Feb 2009) | 9
lines Don't have the Via header stored as a stringfield as it can
change often during the lifetime of a dialog. This issue crept up
with subscriptions on the AA50. When an outgoing NOTIFY is sent a
new branch value is created and the Via header is changed to
reflect it. Since this was a stringfield a new spot in the pool
was used for the value while the old was left untouched/unused.
If the current pool was full a new pool was created. This would
cause memory usage to increase steadily. (issue #AA50-2332)
........
2009-02-16 02:54 +0000 [r175983] Russell Bryant <russell@digium.com>
* main/channel.c: Make the causes array static, and remove the type
name as it is not needed.
2009-02-16 00:26 +0000 [r175952] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_unistim.c, /, channels/chan_sip.c,
include/asterisk/manager.h, doc/unistim.txt: Merged revisions
175921 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r175921 | mvanbaak | 2009-02-16 00:37:03 +0100 (Mon, 16 Feb 2009)
| 3 lines fix mis-spelling of the word registered. Reported by
De_Mon on #asterisk-dev. ........
2009-02-15 21:27 +0000 [r175829-175882] Russell Bryant <russell@digium.com>
* include/asterisk/sched.h, main/sched.c: Make ast_sched_report()
and ast_sched_dump() thread safe.
* channels/chan_sip.c, include/asterisk/sched.h, main/sched.c: Fix
a number of problems with ast_sched_report(). 1) It had numerous
coding guidelines violations with regards to formatting. 2) It
allocated memory using ast_calloc() that was never freed. 3) It
didn't check for failure from the allocation. 4) It used
sprintf() and strcat() to build the result, doing zero checking
to prevent writing past the end of the provided buffer. The
function also lacks API documentation, but that has not been
addressed in this commit.
2009-02-15 20:39 +0000 [r175783-175827] Olle Johansson <oej@edvina.net>
* formats/format_ilbc.c, /: Merged revisions 175825 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r175825 | oej | 2009-02-15 21:33:17 +0100 (Sön, 15 Feb
2009) | 2 lines format_ilbc does not depend on codec libraries
and can therefore always be made. My mistake. Ursäkta! ........
* formats/format_ilbc.c, /: Merged revisions 175792 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r175792 | oej | 2009-02-15 21:20:21 +0100 (Sön, 15 Feb
2009) | 2 lines Disable format_ilbc.so by default, like
codec_ilbc.so ........
* /, channels/chan_sip.c: Merged revisions 175777 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r175777 | oej | 2009-02-15 20:48:38 +0100 (Sön, 15 Feb 2009) | 2
lines Make sure that the debug line is not printed on debug level
0 ........
2009-02-13 20:57 +0000 [r175655-175663] Mark Michelson <mmichelson@digium.com>
* doc/manager_1_1.txt, CHANGES, apps/app_queue.c: Merge queue-reset
branch to Asterisk From a user point-of-view, this adds new CLI
commands and Manager Actions to better facilitate the reloading
of queues and the resetting of their statistics. The new CLI
commands are the "queue reload" and "queue reset stats" commands.
The new manager actions are the QueueReload and QueueReset
commands. Review: http://reviewboard.digium.com/r/115
* doc/manager_1_1.txt, apps/app_chanspy.c: Add manager events for
chanspy starting or stopping (closes issue #14469) Reported by:
caio1982 Patches: chanspy_events2.diff uploaded by caio1982
(license 22)
2009-02-13 20:26 +0000 [r175623-175636] Russell Bryant <russell@digium.com>
* res/res_jabber.c: fix a few more XML documentation problems
* main/pbx.c: add missing </para>
2009-02-13 20:11 +0000 [r175597] David Vossel <dvossel@digium.com>
* configs/iax.conf.sample, channels/iax2.h, channels/chan_iax2.c:
Fixed iax2 key rotation backwards compatibility Turns key
rotation back on by default. Added bit into encryption IE to
indicate whether or not key rotation is supported or not. If it
is not supported then it is not enabled, which insures backwards
compatibility. This eliminates the need for the keyrotate option
in iax.conf, so it has been removed. Review:
http://reviewboard.digium.com/r/159/
2009-02-13 19:49 +0000 [r175591] Mark Michelson <mmichelson@digium.com>
* /, apps/app_voicemail.c: Merged revisions 175590 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r175590 | mmichelson | 2009-02-13 13:47:48 -0600 (Fri,
13 Feb 2009) | 16 lines Fix a potential crash situation when
using IMAP voicemail If calling into VoiceMailMain when using
IMAP storage, it was possible to crash Asterisk by hanging up the
phone when prompted for a voicemail mailbox. This patch fixes the
issue. While it may appear that this patch is superficial, it
allows code execution to continue to the failure case just below
the IMAP_STORAGE code block where this patch has been applied
(closes issue #14473) Reported by: dwpaul Patches:
voicemail_imap_crash_no_mailbox.patch uploaded by dwpaul (license
689) ........
2009-02-13 16:41 +0000 [r175549] Joshua Colp <jcolp@digium.com>
* apps/app_record.c: Add an option to keep the recorded file upon
hangup. (closes issue #14341) Reported by: fnordian
2009-02-13 13:41 +0000 [r175508-175512] Kevin P. Fleming <kpfleming@digium.com>
* CHANGES: document G.722.1/.1C support
* main/frame.c, channels/chan_sip.c, include/asterisk/rtp.h,
channels/chan_h323.c, include/asterisk/frame.h,
formats/format_siren14.c (added), main/rtp.c,
formats/format_siren7.c (added): Add basic (passthrough,
playback, record) support for ITU G.722.1 and G.722.1C (also
known as Siren7 and Siren14) This patch adds passthrough, file
recording and file playback support for the codecs listed above,
with negotiation over SIP/SDP supported. Due to Asterisk's
current limitation of treating a codec/bitrate combination as a
unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are
supported. Along the way, some related work was done: 1) The
rtpPayloadType structure definition, used as a return result for
an API call in rtp.h, was moved from rtp.c to rtp.h so that the
API call was actually usable. The only previous used of the API
all was chan_h323.c, which had a duplicate of the structure
definition instead of doing it the right way. 2) The hardcoded
SDP sample rates for various codecs in chan_sip.c were removed,
in favor of storing these sample rates in rtp.c along with the
codec definitions there. A new API call was added to allow
retrieval of the sample rate for a given codec. 3) Some basic
'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip
*must* decline any media streams offered for these codecs that
are not at the bitrates that we support (otherwise Bad Things
(TM) would result). Review: http://reviewboard.digium.com/r/158/
2009-02-13 04:22 +0000 [r175411-175475] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
* CHANGES: add 'faxbuffers' configuration option information to
CHANGES
* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Add
dynamic fax buffer configuration option to chan_dahdi.conf When
the 'faxdetect' configuration option is used, one may also want
to use the 'faxbuffers' configuration option in chan_dahdi.conf.
This option will dynamically use the configured 'faxbuffers'
buffer policy on a channel for the life of the call following the
detection of fax tones. The faxbuffers buffer policy will be
reverted during call teardown. An example use of 'faxbuffers' is
below. This example would switch to using 6 buffers with a full
buffer policy. faxbuffers=>6,full
2009-02-12 21:41 +0000 [r175368] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Remove useless string copy, and make sscanf
safe again
2009-02-12 21:27 +0000 [r175344] David Vossel <dvossel@digium.com>
* configs/iax.conf.sample, CHANGES, channels/chan_iax2.c: Adds
force encryption option to iax.conf This patch adds
forceencryption=yes as an iax.conf option. When force encryption
is enabled, no unencrypted connections are allowed. This insures
all connections are encrypted. This is a new feature, so CHANGES
and iax.conf.sample are updated as well. (closes issue #13285)
Reported by: sgofferj Tested by: russell Review:
http://reviewboard.digium.com/r/150/
2009-02-12 21:25 +0000 [r175334] Tilghman Lesher <tlesher@digium.com>
* main/udptl.c, /: Merged revisions 175311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r175311 | tilghman | 2009-02-12 15:19:40 -0600 (Thu, 12 Feb 2009)
| 9 lines Fix crashes when receiving certain T.38 packets. Also,
increase the maximum size of T.38 packets and warn users when
they try to set the limits above those maximums. (closes issue
#13050) Reported by: schern Patches: 20090212__bug13050.diff.txt
uploaded by Corydon76 (license 14) Tested by: schern ........
2009-02-12 20:48 +0000 [r175298] Jeff Peeler <jpeeler@digium.com>
* /, main/features.c: Merged revisions 175294 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r175294 | jpeeler | 2009-02-12 14:34:36 -0600 (Thu, 12 Feb 2009)
| 9 lines Fix ParkedCall event information for From field in the
case of a blind transfer If the parker information can not be
obtained from the peer, try and see if the BLINDTRANSFER channel
variable has been set. Previously, a blind transfer to the
ParkAndAnnounce app would return nothing for the From. Closes
AST-189 ........
2009-02-12 20:45 +0000 [r175255-175295] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Avoid using ast_strdupa() in a loop.
* build_tools/cflags.xml: Don't enable something by default that
has a dependency on something _not_ enabled by default.
menuselect was not happy with this.
2009-02-12 18:48 +0000 [r175250] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_iax2.c: correct warning message to not refer
specifically to DAHDI
2009-02-12 18:00 +0000 [r175188] Jeff Peeler <jpeeler@digium.com>
* /, main/features.c: Merged revisions 175187 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r175187 | jpeeler | 2009-02-12 11:57:10 -0600 (Thu, 12 Feb 2009)
| 6 lines Fix crash in event of failed attempt to transfer to
parking The peer may not necessarily exist, such as in the case
of a transfer to ParkAndAnnounce. In this case don't try to play
a sound to it. ........
2009-02-12 17:07 +0000 [r175127] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c: Setting key rotation to be off by default
Key rotation breaks compatibility between (trunk/1.6.1) and
(1.2/1.4/1.6.0). As a follow up to this, I am investigating
possible ways to allow key rotation to be on by default and not
affect the other branches, but for now it must be turned off.
2009-02-12 16:57 +0000 [r175125] Russell Bryant <russell@digium.com>
* /, main/rtp.c: Merged revisions 175124 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r175124 | russell | 2009-02-12 10:51:13 -0600 (Thu, 12 Feb 2009)
| 27 lines Don't send DTMF for infinite time if we do not receive
an END event. I thought that this was going to end up being a
pretty gnarly fix, but it turns out that there was actually
already a configuration option in rtp.conf, dtmftimeout, that was
intended to handle this situation. However, in between Asterisk
1.2 and Asterisk 1.4, the code that processed the option got
lost. So, this commit brings it back to life. The default timeout
is 3 seconds. However, it is worth noting that having this be
configurable at all is not really the recommended behavior in RFC
2833. From Section 3.5 of RFC 2833: Limiting the time period of
extending the tone is necessary to avoid that a tone "gets
stuck". Regardless of the algorithm used, the tone SHOULD NOT be
extended by more than three packet interarrival times. A slight
extension of tone durations and shortening of pauses is generally
harmless. Three seconds will pretty much _always_ be far more
than three packet interarrival times. However, that behavior is
not required, so I'm going to leave it with our legacy behavior
for now. Code from svn/asterisk/team/russell/issue_14460 (closes
issue #14460) Reported by: moliveras ........
2009-02-12 16:28 +0000 [r175121] Mark Michelson <mmichelson@digium.com>
* include/asterisk/astobj2.h, main/astobj2.c: Make lock information
for ao2_trylock be more useful and gnarly Core show locks
information involving an ao2_trylock did not show the function
that called ao2_trylock, but would instead show ao2_trylock as
the source of the lock. This is not useful when trying to debug
locking issues. One bizarre note is that this logic is already in
1.4 but somehow did not get merged to trunk or the 1.6.X
branches.
2009-02-12 14:25 +0000 [r175058-175089] Philippe Sultan <philippe.sultan@gmail.com>
* channels/chan_gtalk.c: Issue a warning message if our candidate's
IP is the loopback address. (closes issue #13985) Reported by:
jcovert Tested by: phsultan
* /, channels/chan_gtalk.c: Merged revisions 175029 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r175029 | phsultan | 2009-02-12 11:16:21 +0100 (Thu, 12
Feb 2009) | 12 lines Set the initiator attribute to lowercase in
our replies when receiving calls. This attribute contains a JID
that identifies the initiator of the GoogleTalk voice session.
The GoogleTalk client discards Asterisk's replies if the
initiator attribute contains uppercase characters. (closes issue
#13984) Reported by: jcovert Patches: chan_gtalk.2.patch uploaded
by jcovert (license 551) Tested by: jcovert ........
2009-02-11 23:12 +0000 [r174945-174951] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Fix a bit of odd logic for announcing position.
Sync with 1.6.0's logic
* apps/app_queue.c: Fix odd "thank you" sound playing behavior in
app_queue.c If someone has configured the queue to play an
position or holdtime announcement, then it is odd and potentially
unexpected to hear a "Thank you for your patience" sound when no
position or holdtime was actually announced. This fixes the
announcement so that the "thanks" sound is only played in the
case that a position or holdtime was actually announced. There is
a way that the "thank you" sound can be played without a position
or holdtime, and that is to set announce-frequency to a value but
keep announce-position and announce-holdtime both turned off.
(closes issue #14227) Reported by: caspy Patches: 14227_v3.patch
uploaded by putnopvut (license 60) Tested by: caspy
* apps/app_dial.c, main/channel.c, main/pbx.c, apps/app_dictate.c,
apps/app_waitforsilence.c, include/asterisk/channel.h: Fix 'd'
option for app_dial and add new option to Answer application The
'd' option would not work for channel types which use RTP to
transport DTMF digits. The only way to allow for this to work was
to answer the channel if we saw that this option was enabled. I
realized that this may cause issues with CDRs, specifically with
giving false dispositions and answer times. I therefore modified
ast_answer to take another parameter which would tell if the CDR
should be marked answered. I also extended this to the Answer
application so that the channel may be answered but not CDRified
if desired. I also modified app_dictate and app_waitforsilence to
only answer the channel if it is not already up, to help not
allow for faulty CDR answer times. All of these changes are going
into Asterisk trunk. For 1.6.0 and 1.6.1, however, all the
changes except for the change to the Answer application will go
in since we do not introduce new features into stable branches
(closes issue #14164) Reported by: DennisD Patches: 14164.patch
uploaded by putnopvut (license 60) Tested by: putnopvut Review:
http://reviewboard.digium.com/r/145
2009-02-11 14:44 +0000 [r174844] Joshua Colp <jcolp@digium.com>
* main/channel.c: Tell the device state core a change happened when
a channel is freed but not a specific state. We need to do this
because while we know that the freeing of the channel may cause
something to become not in use we do not know this for sure.
There may be another channel that is still up which would cause
it to be in use. (closes issue #13238) Reported by: kowalma
Patches: 20090121__bug13238.diff.txt uploaded by Corydon76
(license 14) Tested by: alecdavis
2009-02-10 23:17 +0000 [r174764-174805] Mark Michelson <mmichelson@digium.com>
* apps/app_chanspy.c: Fix potential for stack overflows in
app_chanspy.c When using the 'g' or 'e' options, the stack
allocations that were used could cause a stack overflow if a
spyer stayed on the line long enough without actually
successfully spying on anyone. The problem has been corrected by
using static buffers and copying the contents of the appropriate
strings into them instead of using functions like alloca or
ast_strdupa
* main/manager.c: Fix an fd leak that would occur in HTTP AMI
sessions The explanation behind this fix is a bit complicated,
and I've already typed it up in the code as a huge comment inside
of manager.c, so I'll give the abridged version here. We needed a
way to separate action-specific data from session-specific data.
Unfortunately, the only way to maintain API compatibility and to
not have to change every single manager action was to rename the
current mansession structure and wrap it inside a new mansession
structure which actually contains action- specific data. (closes
issue #14364) Reported by: awk Patches: 14364_better.patch
uploaded by putnopvut (license 60) Tested by: putnopvut Review:
http://reviewboard.digium.com/r/148/
2009-02-10 20:15 +0000 [r174710] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Only decrease inringing count if above zero.
(issue #13238) Reported by: kowalma
2009-02-10 19:38 +0000 [r174705] Kevin P. Fleming <kpfleming@digium.com>
* main/slinfactory.c, include/asterisk/slinfactory.h: improve
slinfactory API to remove implicit sample rate and require
explicit sample rate selection by creator of the slinfactory
2009-02-10 18:16 +0000 [r174584] Matthew Nicholson <mnicholson@digium.com>
* /, main/jitterbuf.c: Merged revisions 174583 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r174583 | mnicholson | 2009-02-10 11:52:42 -0600 (Tue, 10 Feb
2009) | 18 lines Improve behavior of jitterbuffer when
maxjitterbuffer is set. This change improves the way the
jitterbuffer handles maxjitterbuffer and dramatically reduces the
number of frames dropped when maxjitterbuffer is exceeded. In the
previous jitterbuffer, when maxjitterbuffer was exceeded, all new
frames were dropped until the jitterbuffer is empty. This change
modifies the code to only drop frames until maxjitterbuffer is no
longer exceeded. Also, previously when maxjitterbuffer was
exceeded, dropped frames were not tracked causing stats for
dropped frames to be incorrect, this change also addresses that
problem. (closes issue #14044) Patches: bug14044-1.diff uploaded
by mnicholson (license 96) Tested by: mnicholson Review:
http://reviewboard.digium.com/r/144/ ........
2009-02-10 17:48 +0000 [r174543-174580] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Set the type for the peer structure to be a
peer as the default. (closes issue #14447) Reported by: triccyx
* channels/chan_sip.c: Make the logic for inuse and inringing
manipluation match that of 1.4. The old broken logic would reset
the values back to 0 during certain scenarios causing the wrong
state to be reported. (closes issue #14399) Reported by: caspy
(issue #13238) Reported by: kowalma
2009-02-10 07:06 +0000 [r174470-174503] Tilghman Lesher <tlesher@digium.com>
* apps/app_stack.c, apps/app_voicemail.c: Fix0ring build
* apps/app_stack.c: Remove the usage of the KeepAlive app, as it no
longer exists.
2009-02-10 04:49 +0000 [r174370-174435] Steve Murphy <murf@digium.com>
* apps/app_rpt.c: This patch removes the use of AST_PBX_KEEPALIVE
from app_rpt.c. (closes issue #14435) Reported by: D_McNaul
* apps/app_rpt.c: More intptr_t work.
* /, apps/app_rpt.c: Merged revisions 174369 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r174369 | murf | 2009-02-09 19:27:40 -0700 (Mon, 09 Feb 2009) | 5
lines This patch solves some compiler complaints in both 32 and
64-bit environments. ........
2009-02-09 17:27 +0000 [r174327] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Fix something I messed up in the merge I
just did
2009-02-09 17:26 +0000 [r174325] David Vossel <dvossel@digium.com>
* apps/app_externalivr.c: Fixes issue with hangups not being sent
and external process never terminating. The ignore_hangup,
run_dead, and noanswer flags were never initilized to zero
causing hangups to never be issued. If the external script
expects to be notified of a hangup and never receives one, it
runs indefinitely. (closes issue #14251) Reported by: chris-mac
Tested by: dvossel
2009-02-09 17:20 +0000 [r174301] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 174282 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r174282 | mmichelson | 2009-02-09 11:11:05 -0600 (Mon, 09 Feb
2009) | 12 lines Don't do an SRV lookup if a port is specified
RFC 3263 says to do A record lookups on a hostname if a port has
been specified, so that's what we're going to do. See section
4.2. (closes issue #14419) Reported by: klaus3000 Patches:
patch_chan_sip_nosrvifport_1.4.23.txt uploaded by klaus3000
(license 65) ........
2009-02-09 14:49 +0000 [r174219] Joshua Colp <jcolp@digium.com>
* /, res/res_musiconhold.c: Merged revisions 174218 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r174218 | file | 2009-02-09 10:48:21 -0400 (Mon, 09 Feb
2009) | 4 lines Don't overwrite our pointer to the music class
when music on hold stops. We will use this if it starts again to
see if we can resume the music where it left off. (closes issue
#14407) Reported by: mostyn ........
2009-02-07 16:16 +0000 [r174149] Russell Bryant <russell@digium.com>
* /, res/snmp/agent.c: Merged revisions 174148 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r174148 | russell | 2009-02-07 10:15:07 -0600 (Sat, 07 Feb 2009)
| 2 lines Fix a race condition that could cause a crash. ........
2009-02-06 23:51 +0000 [r174084] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
* /, channels/chan_sip.c: Merged revisions 174082 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r174082 | dhubbard | 2009-02-06 17:36:03 -0600 (Fri, 06 Feb 2009)
| 5 lines check ast_strlen_zero() before calling ast_strdupa() in
sip_uri_headers_cmp() and sip_uri_params_cmp() The reporter
didn't actually upload a properly-formed patch, instead a
modified chan_sip.c file was uploaded. I created a patch to
determine the changes, then modified the suggested changes to
create a proper fix. The summary above is a complete description
of the changes. (closes issue #13547) Reported by: tecnoxarxa
Patches: chan_sip.c.gz uploaded by tecnoxarxa (license 258)
Tested by: tecnoxarxa ........
2009-02-06 20:12 +0000 [r174046] David Vossel <dvossel@digium.com>
* configs/iax.conf.sample, CHANGES, channels/chan_iax2.c: Adds
immediate yes/no option to iax.conf This is very similar to the
DAHDI immediate=yes option. When the phone is picked up, instead
of giving a dialtone it connects directly to the "s" extension.
Changes where implemented in chan_iax2.c to directly connect to
the "s" extension in the appropriate context when this option is
enabled. Examples explaining its use are added to
iax2.conf.sample. CHANGES has been updated as well. (closes issue
#14266) Reported by: jcovert Patches: chan_iax2.c.patch-trunk
uploaded by jcovert (license 551) iax.conf.sample.patch uploaded
by jcovert (license 551) Tested by: jcovert, dvossel Review:
http://reviewboard.digium.com/r/143/
2009-02-06 19:28 +0000 [r173974-174041] Joshua Colp <jcolp@digium.com>
* channels/chan_dahdi.c: Don't subscribe to a mailbox on pseudo
channels. It is futile. This solves an issue where duplicated
pseudo channels would cause a crash because the first one would
unsubscribe and the next one would also try to unsubscribe the
same subscription. (closes issue #14322) Reported by: amessina
* /, channels/chan_sip.c: Merged revisions 173967-173968 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r173967 | file | 2009-02-06 13:14:15 -0400 (Fri, 06 Feb 2009) | 4
lines Some clients do not put the call-id for replaces at the
beginning, so support it being anywhere in the string. (closes
issue #14350) Reported by: fhackenberger ........ r173968 | file
| 2009-02-06 13:15:01 -0400 (Fri, 06 Feb 2009) | 2 lines Remove a
debug message I put in by accident. ........
2009-02-06 16:28 +0000 [r173952] Matthew Nicholson <mnicholson@digium.com>
* /, channels/chan_sip.c: Merged revisions 173917 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r173917 | mnicholson | 2009-02-06 10:20:23 -0600 (Fri, 06 Feb
2009) | 7 lines Limit the addition of the Contact header in SIP
responses according to various SIP RFCs. (closes issue #13602)
Reported by: hjourdain Tested by: mnicholson ........
2009-02-06 15:59 +0000 [r173902] Joshua Colp <jcolp@digium.com>
* main/audiohook.c, apps/app_chanspy.c: Always detach and destroy
the whisper and barge audiohooks. Additionally also allow an
audiohook to be detached if it has not been attached. (closes
issue #14414) Reported by: bluecrow76
2009-02-06 10:55 +0000 [r173848-173858] Russell Bryant <russell@digium.com>
* include/asterisk/sched.h, channels/chan_iax2.c, main/sched.c: Add
a common implementation of a scheduler context with a dedicated
thread. This commit expands the Asterisk scheduler API to include
a common implementation of a scheduler context being processed by
a dedicated thread. chan_iax2 has been updated to use this new
code. Also, as a result, this resolves some race conditions
related to the previous chan_iax2 scheduler handling. Related to
rev 171452 which resolved the same issues in 1.4. Code from
team/russell/sched_thread2 Review:
http://reviewboard.digium.com/r/129/
* main/manager.c: Resolve a memory leak that would occur on an
invalid channel given to Action: Status
2009-02-05 23:48 +0000 [r173773-173776] Mark Michelson <mmichelson@digium.com>
* configs/extensions.conf.sample: Update extensions.conf.sample to
be correct. In trunk, the only necessary change pointed out was
that the call to ChanIsAvail uses an option that has been
removed. For the 1.6.1 branch, however, it appears that the
sample file is badly in need of updating since there are |'s used
all over the place there. My tentative plan is just to copy
trunk's sample config file to those branches since the info there
is most up-to-date and should be correct for use in 1.6.1 Thanks
to macli in #asterisk-dev for bringing this up
* apps/app_voicemail.c: Properly set "seen" and "unseen" flags when
moving messages from the new to the old folder when using IMAP
for voicemail storage (closes issue #13905) Reported by: jaroth
Patches: foldermove_v2.patch uploaded by jaroth (license 50)
2009-02-05 21:00 +0000 [r173697] Jeff Peeler <jpeeler@digium.com>
* /, apps/app_voicemail.c: Merged revisions 173696 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r173696 | jpeeler | 2009-02-05 14:47:51 -0600 (Thu, 05
Feb 2009) | 12 lines Add new configuration option to make shared
IMAP mailboxes function as expected. The new option is
"imapvmshareid" which is an ID to tag multiple mailboxes using
the same IMAP storage location to function as one mailbox. This
allows all messages to be retrieved for any user in the group.
The patch alters the 'X-Asterisk-VM-Extension' header that is
responsible for matching voicemails for a given user. (closes
issue #13673) Reported by: howardwilkinson ........
2009-02-05 20:30 +0000 [r173693] Mark Michelson <mmichelson@digium.com>
* /, apps/app_queue.c: Merged revisions 173692 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r173692 | mmichelson | 2009-02-05 14:29:09 -0600 (Thu, 05 Feb
2009) | 12 lines Fix situations where queue members could be
autopaused unexpectedly Specifically, this patch prevents us from
autopausing members when we receive a busy or congestion frame
from them. (closes issue #14376) Reported by: fiddur Patches:
14376.patch uploaded by putnopvut (license 60) Tested by: fiddur
........
2009-02-05 19:36 +0000 [r173657] Tilghman Lesher <tlesher@digium.com>
* res/res_config_sqlite.c: Change the first field, or we don't get
the necessary field separation.
2009-02-05 18:48 +0000 [r173507-173593] Mark Michelson <mmichelson@digium.com>
* /, apps/app_mixmonitor.c: Merged revisions 173592 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r173592 | mmichelson | 2009-02-05 12:47:24 -0600 (Thu,
05 Feb 2009) | 3 lines Add some missing cleanup to app_mixmonitor
........
* /, apps/app_mixmonitor.c: Merged revisions 173559 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r173559 | mmichelson | 2009-02-05 11:34:33 -0600 (Thu,
05 Feb 2009) | 25 lines Fix a problem where a channel pointer
becomes invalid due to masquerading or hanging up. app_mixmonitor
runs its own thread to monitor the channel's activity and write
the mixed audio to a file. Since this thread runs independently
of the channel, it is possible that the mixmonitor thread's
channel pointer will point to freed memory when the channel
either is masqueraded or hangs up (technically, both cases are
hangups, but we need to handle the cases slightly differently).
The solution for this is to employ a datastore, which has the
nice benefit of allowing us to hook into channel masquerades and
hangups and update our pointer as necessary. If this looks
familiar, this same technique is employed in app_chanspy.
app_chanspy is a bit more involved since it does a lot more
operations on the channel that is being spied upon.
app_mixmonitor does have an extra touch that app_chanspy doesn't
have, though. Since there is a thread race between the channel's
thread and the mixmonitor thread on a hangup, we em- ploy a
condition-and-boolean combination to ensure that the channel
thread finishes with our structure before the mixmonitor thread
attempts to free it. No crashes! (closes issue #14374) Reported
by: aragon Patches: 14374.patch uploaded by putnopvut (license
60) Tested by: aragon, putnopvut ........
* apps/app_queue.c: Fix some areas where the incorrect interface
was passed to ast_device_state I swear it feels like I already
did this once... (closes issue #14359) Reported by: francesco_r
2009-02-04 21:26 +0000 [r173503] Tilghman Lesher <tlesher@digium.com>
* res/res_jabber.c: Add XML documentation for the applications and
functions in res_jabber (closes issue #14405) Reported by: snuffy
Patches: xml_jabber.diff uploaded by snuffy (license 35)
2009-02-04 21:25 +0000 [r173502] David Vossel <dvossel@digium.com>
* channels/iax2-parser.h, channels/chan_iax2.c: Fixes issue with
IAX2 transfer not handing off calls. Reverts changes in 116884
Fixes issue with IAX2 transfers not taking place. As it was, a
call that was being transfered would never be handed off
correctly to the call ends because of how call numbers were
stored in a hash table. The hash table, "iax_peercallno_pvt",
storing all the current call numbers did not take into account
the complications associated with transferring a call, so a
separate hash table was required. This second hash table
"iax_transfercallno_pvt" handles calls being transfered, once the
call transfer is complete the call is removed from the transfer
hash table and added to the peer hash table resuming normal
operations. Addition functions were created to handle storing,
removing, and comparing items in the iax_transfercallno_pvt
table. The changes reverted in 116884 caused backwards
compatibility issues involving iax2 transfer with 1.6.0, 1.4, and
1.2. (closes issue #13468) Reported by: nicox Tested by: dvossel
2009-02-04 21:17 +0000 [r173500] Jeff Peeler <jpeeler@digium.com>
* /, main/features.c, include/asterisk/features.h: Merged revisions
173211 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r173211 | jpeeler | 2009-02-03 15:57:01 -0600 (Tue, 03 Feb 2009)
| 17 lines Parking attempts made to one end of a bridge no longer
will hang up due to a parking failure. Parking attempts made
using either one-touch, or doing either a blind or assisted
transfer to the parking extension now keep up the bridge instead
of hanging up the attempted parked party. Normal causes for the
parking attempt to fail includes the specific specified extension
(via PARKINGEXTEN) not being available or if all the parking
spaces are currently in use. To avoid having to reverse a
masquerade park_space_reserve was made to provide foresight if a
parking attempt will succeed and if so reserve the parking space.
(closes issue #13494) Reported by: mdu113 Reviewed by Russell:
http://reviewboard.digium.com/r/133/ ........
2009-02-04 18:48 +0000 [r173458] Tilghman Lesher <tlesher@digium.com>
* main/tcptls.c: When using a socket as a FILE *, the stdio
functions will sometimes try to do an fseek() on the stream,
which is an invalid operation for a socket. Turning off buffering
explicitly lets the stdio functions know they cannot do this,
thus avoiding a potential error. (closes issue #14400) Reported
by: fnordian Patches: tcptls.patch uploaded by fnordian (license
110)
2009-02-04 17:45 +0000 [r173354-173397] Mark Michelson <mmichelson@digium.com>
* /, apps/app_chanspy.c: Merged revisions 173396 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r173396 | mmichelson | 2009-02-04 11:44:48 -0600 (Wed, 04 Feb
2009) | 3 lines Revert my previous change because it was stupid
........
* /, apps/app_chanspy.c: Merged revisions 173392 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r173392 | mmichelson | 2009-02-04 11:40:29 -0600 (Wed, 04 Feb
2009) | 3 lines Add a missing unlock. Extremely unlikely to ever
matter, but it's needed. ........
* main/file.c: Fix a problem where file playback would cause fds to
remain open forever The problem came from the fact that a frame
read from a format interpreter was not freed. Adding a call to
ast_frfree fixed this. The explanation for why this caused the
problem is a bit complex, but here goes: There was a problem in
all versions of Asterisk where the embedded frame of a filestream
structure was referenced after the filestream was freed. This was
fixed by adding reference counting to the filestream structure.
The refcount would increase every time that a filestream's frame
pointer was pointing to an actual frame of data. When the frame
was freed, the refcount would decrease. Once the refcount reached
0, the filestream was freed, and as part of the operation, the
open files were closed as well. Thus it becomes more clear why a
missing ast_frfree would cause a reference leak and cause the
files to not be closed. You may ask then if there was a frame
leak before this patch. The answer to that is actually no! The
filestream code was "smart" enough to know that since the frame
we received came from a format interpreter, the frame had no
malloced data and thus didn't need to be freed. Now, however,
there is cleanup that needs to be done when we finish with the
frame, so we do need to call ast_frfree on the frame to be sure
that the refcount for the filestream is decremented
appropriately. (closes issue #14384) Reported by: fiddur Patches:
14384.patch uploaded by putnopvut (license 60) Tested by: fiddur,
putnopvut
2009-02-04 00:43 +0000 [r173311] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c, pbx/pbx_config.c: Ensure that commas placed in the
middle of extension character classes do not interfere with
correct parsing of the extension. Also, if an unterminated
character class DOES make its way into the pbx core (through some
other method), ensure that it does not crash Asterisk. (closes
issue #14362) Reported by: Nick_Lewis Patches:
20090129__bug14362.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76
2009-02-03 17:35 +0000 [r173169] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: Broke up the large conditional blocks so
it is easy to see if a function is compiled.
2009-02-03 00:29 +0000 [r173104-173130] Tilghman Lesher <tlesher@digium.com>
* configure, include/asterisk/autoconfig.h.in, configure.ac,
main/xml.c, include/asterisk/compiler.h, apps/app_stack.c,
include/asterisk/optional_api.h: 1. Make OS X compile cleanly
with app_stack. 2. Use curl to download sound files, as curl is
installed natively on OS X, whereas wget and fetch are not.
(closes issue #14332) Reported by: oej Tested by: Corydon76
* /, configs/extensions.conf.sample: Merged revisions 173070 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r173070 | tilghman | 2009-02-02 18:15:59 -0600 (Mon, 02 Feb 2009)
| 5 lines Add warning to standard config, that globals may be
overridden by other dialplan configuration files. (closes issue
#14388) Reported by: macli ........
2009-02-02 23:57 +0000 [r173067] Terry Wilson <twilson@digium.com>
* /, main/features.c: Merged revisions 173066 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r173066 | twilson | 2009-02-02 17:48:06 -0600 (Mon, 02 Feb 2009)
| 2 lines Fix a feature inheritance bug I added after code review
........
2009-02-02 23:21 +0000 [r173028-173047] Mark Michelson <mmichelson@digium.com>
* main/manager.c, CHANGES: Reverting commit number 173028 as there
are some potential issues
* main/manager.c, CHANGES: Add a CLI command to log out a manager
user (closes issue #13877) Reported by: eliel Patches:
cli_manager_logout.patch.txt uploaded by eliel (license 64)
Tested by: eliel, putnopvut
2009-02-02 20:40 +0000 [r172963] Richard Mudgett <rmudgett@digium.com>
* /: Recorded merge of revisions 172962 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r172962 | rmudgett | 2009-02-02 14:28:54 -0600 (Mon, 02 Feb 2009)
| 11 lines channels/chan_dahdi.c * Added doxygen comments to the
major dahdi structures. * Fixed PRI using an incorrect string
value if the extension delimiter is not present in the Dial()
function. * Fixed some uninitialized string variables on FXS
ports. configs/chan_dahdi.conf.sample * Updated some
documentation. These changes are already in trunk -r172400
........
2009-02-02 19:02 +0000 [r172929] Steve Murphy <murf@digium.com>
* apps/app_dial.c, main/features.c, CHANGES,
include/asterisk/features.h: This reverts the changes I made for
11583; will reviewboard this before committing again... reopened
11583 until all Russell's issues are resolved.
2009-02-02 18:13 +0000 [r172894] Leif Madsen <lmadsen@digium.com>
* configs/res_ldap.conf.sample: Update the res_ldap.conf file with
a better working example. (closes issue #13861) Reported by:
scramatte Patches: __20080110-res_ldap.conf-2.patch uploaded by
blitzrage (license 10) Tested by: jcovert
2009-02-02 17:37 +0000 [r172890] Steve Murphy <murf@digium.com>
* apps/app_dial.c, main/features.c, CHANGES,
include/asterisk/features.h: This change allows the disconnect
feature (as in "one-touch" in features.c) to be used within the
dial app, before a call is bridged. Many thanks to sobomax for
submitting this patch. Quoting from bug 11582: "So the goal of
the patch was to use the user configured feature code during the
call setup phase. The original ast_feature_interpret() function
is not well suited for this purpose as it uses much call bridge
specific data and doesn't separate a detection of feature from a
feature handler call. So a new function ast_feature_detect() has
been extracted off the ast_feature_interpret() function but
keeping the original logic intact except some insignificant
changes to locking. "Having created the ast_feature_detect()
function the possibility to use feature detection in almost any
place of the asterisk code. So a call to this function has been
added to wait_for_answer() function of app_dial.so module. This
code doesn't call the feature handler however and uses old call
leg disconnect logic to make the changes as small and simple as
possible to prevent unexpected problems. A disconnect feature
currently is the only one supported during call setup as other
features as call parking and call transfer don't make much sense
during call setup. However if need in some of the features would
arise it is much easier to implement as the infrastructure
changes are already in place with this patch." I have cleaned up
the patch somewhat, and verified that the existing functionality
is not harmed, and that the new functionality works. Terry has
committed his stuff, and there were no conflicts (see 14274).
(closes issue #11583) Reported by: sobomax Patches:
patch-apps__app_dial.c uploaded by sobomax (license 359)
patch-include__asterisk__features.h uploaded by sobomax (license
359) patch-res__res_features.c uploaded by sobomax (license 359)
enable-features-during-call-setup.diff uploaded by sobomax
(license 359) 11583.newdiff uploaded by murf (license 17)
enable-features-during-call-setup-1.diff uploaded by sobomax
(license 359) 11583.latest-patch uploaded by murf (license 17)
Tested by: sobomax, murf
2009-02-02 16:42 +0000 [r172855] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Fix a spelling mistake.
2009-02-02 10:46 +0000 [r172816-172818] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Add a todo. I do need to really check what's
going on with this kill-the-user business ;-) Why do we suddenly
have two flags to set peer type?
* channels/chan_sip.c: Small formatting change
* channels/chan_sip.c: Add some well-needed improvements to the
wishlist in the code, so that we can close some bug reports.
2009-02-02 01:41 +0000 [r172778] Sean Bright <sean.bright@gmail.com>
* channels/chan_sip.c: The CID lookup feature wasn't actually
working properly with dialog-info+xml supporting devices. The
devices (snoms, specifically) need to receive a SIP URI instead
of just an extension. This adds that functionality.
2009-02-01 02:44 +0000 [r172706-172741] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: Blank argument crashes Asterisk (closes
issue #14377) Reported by: amorsen
* funcs/func_strings.c: Don't increment the loop, now that
incrementing is taken care of by the decoder function. (closes
issue #14363) Reported by: andrew53 Patches:
func_strings_filter.patch uploaded by andrew53 (license 519)
2009-01-30 22:22 +0000 [r172598] Mark Michelson <mmichelson@digium.com>
* include/asterisk/channel.h: Fix redefinition of flag in channel.h
2009-01-30 21:50 +0000 [r172580-172581] Terry Wilson <twilson@digium.com>
* configs/features.conf.sample: Remove incorrect line from sample
config
* apps/app_dial.c, main/global_datastores.c, main/features.c,
include/asterisk/global_datastores.h, CHANGES,
configs/features.conf.sample: Merged revisions 172517 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r172517 | twilson | 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009)
| 37 lines Fix feature inheritance with builtin features When
using builtin features like parking and transfers, the
AST_FEATURE_* flags would not be set correctly for all instances
when either performing a builtin attended transfer, or parking a
call and getting the timeout callback. Also, there was no way on
a per-call basis to specify what features someone should have on
picking up a parked call (since that doesn't involve the Dial()
command). There was a global option for setting whether or not
all users who pickup a parked call should have
AST_FEATURE_REDIRECT set, but nothing for DISCONNECT, AUTOMON, or
PARKCALL. This patch: 1) adds the BRIDGE_FEATURES dialplan
variable which can be set either in the dialplan or with setvar
in channels that support it. This variable can be set to any
combination of 't', 'k', 'w', and 'h' (case insensitive matching
of the equivalent dial options), to set what features should be
activated on this channel. The patch moves the setting of the
features datastores into the bridging code instead of app_dial to
help facilitate this. 2) adds global options parkedcallparking,
parkedcallhangup, and parkedcallrecording to be similar to the
parkedcalltransfers option for globally setting features. 3) has
builtin_atxfer call builtin_parkcall if being transfered to the
parking extension since tracking everything through multiple
masquerades, etc. is difficult and error-prone 4) attempts to fix
all cases of return calls from parking and completed builtin
transfers not having the correct permissions (closes issue
#14274) Reported by: aragon Patches:
fix_feature_inheritence.diff.txt uploaded by otherwiseguy
(license 396) Tested by: aragon, otherwiseguy Review
http://reviewboard.digium.com/r/138/ ........
2009-01-30 18:36 +0000 [r172441-172548] Tilghman Lesher <tlesher@digium.com>
* funcs/func_aes.c: Parameter position reversed in documentation
* /, autoconf/ast_func_fork.m4, configure, main/app.c,
apps/app_rpt.c, main/asterisk.c: Merged revisions 172438 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r172438 | tilghman | 2009-01-29 16:54:29 -0600 (Thu, 29 Jan 2009)
| 9 lines Lose the CAP_NET_ADMIN at every fork, instead of at
startup. Otherwise, if Asterisk runs as a non-root user and the
administrator does a 'restart now', Asterisk loses the ability to
set QOS on packets. (closes issue #14004) Reported by: nemo
Patches: 20090105__bug14004.diff.txt uploaded by Corydon76
(license 14) Tested by: Corydon76 ........
2009-01-29 23:15 +0000 [r172370-172440] Richard Mudgett <rmudgett@digium.com>
* main/cli.c: Remove tabs from comment
* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample:
channels/chan_dahdi.c * Added doxygen comments to the major dahdi
structures. * Fixed PRI and SS7 using an incorrect string value
if the extension delimiter is not present in the Dial() function.
* Fixed SS7 not checking if the dialed extension is at least as
long as the stripmsd option. * Fixed PRI not handling unknown
TON/NPI prefix letters correctly. * Fixed some uninitialized
string variables on FXS ports. configs/chan_dahdi.conf.sample *
Updated some documentation.
* include/asterisk/say.h: Fixed some doxygen comments
2009-01-29 17:10 +0000 [r172318-172319] Olle Johansson <oej@edvina.net>
* channels/chan_local.c: Revert two lines that was extra, but only
on fridays.
* apps/app_dial.c, channels/chan_local.c, channels/chan_sip.c,
include/asterisk/causes.h, apps/app_queue.c: Fix "cancel answered
elsewhere" through app_queue with members in chan_local. Also,
implement a private cause code (as suggested by Tilghman). This
works with chan_sip, but doesn't propagate through chan_local.
2009-01-29 16:48 +0000 [r172315] Tilghman Lesher <tlesher@digium.com>
* configs/func_odbc.conf.sample: Better document mode=multirow,
based upon a conversation with Jared.
2009-01-29 13:47 +0000 [r172271] Leif Madsen <lmadsen@digium.com>
* contrib/scripts/realtime_pgsql.sql: The realtime_pgsql.sql script
is missing a couple of fields. closes issue #14339) Reported by:
fiddur Patches: realtime_pgsql.sql.diff uploaded by fiddur
(license 678)
2009-01-29 13:24 +0000 [r172173-172270] Olle Johansson <oej@edvina.net>
* configs/sip.conf.sample, CHANGES: Update documentation
* include/asterisk/app.h, channels/chan_sip.c, main/app.c: - Make
sure we set setvar= variables on outbound calls too, not only
inbound calls. - Also, change a function in app.c to return a
userful value instead of always returning 0. Patch by fnordian,
changed by Corydon76 and myself. This does not close the bug
report, as fnordian had an additional change we're still
discussing. (related to issue #14059) Reported by: fnordian
Patches: chan_sip_hfield.patch uploaded by fnordian (license 110)
20090116__bug14059.diff.txt uploaded by Corydon76 (license 14)
Tested by: fnordian, Corydon76, oej
* channels/chan_sip.c: Make sure register= line supports both port
and expiry at the same time. (closes issue #14185) Reported by:
Nick_Lewis Patches: chan_sip.c-expiryrequest6.patch uploaded by
Nick (license 657) Tested by: Nick_Lewis
* /, channels/chan_sip.c: Merged revisions 172169 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r172169 | oej | 2009-01-29 09:48:18 +0100 (Tor, 29 Jan 2009) | 16
lines Make sure that we always add the hangupcause headers. In
some cases, the owner was disconnected before we checked for the
cause. This patch implements a temporary storage in the pvt and
use that instead. The code is based on ideas from code from
Adomjan in issue #13385 (Add support for Reason: header) Thanks
to Klaus Darillion for testing! (closes issue #14294) related to
issue #13385 Reported by: klaus3000 and adomjan Patches:
bug14294b.diff uploaded by oej (license 306) Based on
20080829_chan_sip.c-q850reason_header.patch uploaded by adomjan
(license 487) Tested by: oej, klaus3000 ........
2009-01-28 22:52 +0000 [r172132] Steve Murphy <murf@digium.com>
* channels/chan_misdn.c: A further correction: cast the sizeof to
an int.
2009-01-28 22:48 +0000 [r172131] Tilghman Lesher <tlesher@digium.com>
* res/res_config_odbc.c: Fix how we skip fields (to avoid fields
which don't exist) when doing an UPDATE. (closes issue #14205)
Reported by: maxgo Patches: 20090128__bug14205__5.diff.txt
uploaded by Corydon76 (license 14) Tested by: blitzrage
2009-01-28 21:48 +0000 [r172063-172099] Steve Murphy <murf@digium.com>
* channels/chan_misdn.c: my gcc (Ubuntu 4.3.2-1ubuntu11) 4.3.2
didn't like the \%ld and issued a warning, breaking my dev-mode
build. This fixes it.
* apps/app_channelredirect.c, main/pbx.c, main/manager.c, /,
main/features.c, include/asterisk/channel.h: Merged revisions
172030 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r172030 | murf | 2009-01-28 11:51:16 -0700 (Wed, 28 Jan 2009) |
46 lines This patch fixes h-exten running misbehavior in
manager-redirected situations. What it does: 1. A new Flag value
is defined in include/asterisk/channel.h,
AST_FLAG_BRIDGE_HANGUP_DONT, which used as a messenge to the
bridge hangup exten code not to run the h-exten there (nor
publish the bridge cdr there). It will done at the pbx-loop level
instead. 2. In the manager Redirect code, I set this flag on the
channel if the channel has a non-null pbx pointer. I did the same
for the second (chan2) channel, which gets run if name2 is set...
and the first succeeds. 3. I restored the ending of the cdr for
the pbx loop h-exten running code. Don't know why it was removed
in the first place. 4. The first attempt at the fix for this bug
was to place code directly in the async_goto routine, which was
called from a large number of places, and could affect a large
number of cases, so I tested that fix against a fair number of
transfer scenarios, both with and without the patch. In the
process, I saw that putting the fix in async_goto seemed not to
affect any of the blind or attended scenarios, but still, I was
was highly concerned that some other scenarios I had not tested
might be negatively impacted, so I refined the patch to its
current scope, and jmls tested both. In the process, tho, I saw
that blind xfers in one situation, when the one-touch blind-xfer
feature is used by the peer, we got strange h-exten behavior. So,
I inserted code to swap CDRs and to set the HANGUP_DONT field, to
get uniform behavior. 5. I added code to the bridge to obey the
HANGUP_DONT flag, skipping both publishing the bridge CDR, and
running the h-exten; they will be done at the pbx-loop (higher)
level instead. 6. I removed all the debug logs from the patch
before committing. 7. I moved the AUTOLOOP set/reset in the
h-exten code in res_features so it's only done if the h-exten is
going to be run. A very minor performance improvement, but
technically correct. (closes issue #14241) Reported by: jmls
Patches: 14241_redirect_no_bridgeCDR_or_h_exten_via_transfer
uploaded by murf (license 17) Tested by: murf, jmls ........
2009-01-28 17:27 +0000 [r171964] Tilghman Lesher <tlesher@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 171963 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r171963 | tilghman | 2009-01-28 11:25:18 -0600 (Wed, 28
Jan 2009) | 2 lines Clarify log message (suggested by manxpower
on #asterisk-dev) ........
2009-01-28 14:39 +0000 [r171838-171925] Olle Johansson <oej@edvina.net>
* CHANGES: Yep. Documentation is important.
* apps/app_queue.c: Add final part of previously committed work for
answered elsewhere in queue - the missing piece that started with
app_dial() earlier on. This is to avoid having the list and
counter of missed calls being touched by queue calls. Add the C
option to queue() and nothing will be logged on phones that
support the Reason: header on SIP cancel, like the SNOM phones.
* configs/sip.conf.sample: Add some more notes about device
matching.
* /, configs/sip.conf.sample: Merged revisions 171837 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r171837 | oej | 2009-01-28 14:07:27 +0100 (Ons, 28 Jan
2009) | 2 lines Add a better explanation of the difference
between the device namespace and the dialplan for newbies.
........
2009-01-28 00:17 +0000 [r171797] Mark Michelson <mmichelson@digium.com>
* funcs/func_aes.c: Fix some signedness problems in func_aes.c
2009-01-27 23:28 +0000 [r171793] Matthew Fredrickson <creslin@digium.com>
* channels/chan_dahdi.c: Don't complain about lack of D-channels on
PTMP connections
2009-01-27 22:43 +0000 [r171757] David Vossel <dvossel@digium.com>
* funcs/func_aes.c (added), CHANGES: Adding AES_ENCRYPT and
AES_DECRYPT dialplan functions. (closes issue #14301) Reported
by: amorsen review: http://reviewboard.digium.com/r/128/
2009-01-27 21:58 +0000 [r171618-171691] Mark Michelson <mmichelson@digium.com>
* channels/chan_agent.c: Merged revisions 171689 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r171689 | mmichelson | 2009-01-27 15:55:08 -0600 (Tue, 27 Jan
2009) | 39 lines Fix devicestate problems for "always-on" agent
channels A revision to chan_agent attempted to "inherit" the
device state of the underlying channel in order to report the
device state of an agent channel more accurately. The problem
with the logic here is that it makes no sense to use this for
always-on agents. If the agent is logged in, then to the
underlying channel, the agent will always appear to be "in use,"
no matter if the agent is on a call or not. The reason is that to
the underlying channel, the channel is currently in use on a call
to the AgentLogin application. The most common cause that I found
for this issue to occur was for a SIP channel to be the
underlying channel type for an Agent channel. If the SIP phone
re-registers, then the registration will cause the device state
core to query the device state of the SIP channel. Since the SIP
channel is in use, the Agent channel would also inherit this
status. Once the agent channel was set to "in use" there was no
way that the device state could change on that channel unless the
agent logged out. The solution for this problem is a bit
different in 1.4 than it is in the other branches. In 1.4, there
will be a one-line fix to make sure that only callback agents
will inherit device state from their underlying channel type. For
the other branches of Asterisk, since callback support has been
removed, there is also no need for device state inheritance in
chan_agent, so I will simply be removing it from the code. In
addition, the 1.4 source is getting a new comment to help the
next person who edits chan_agent.c. I'm adding a comment that a
agent_pvt's loginchan field may be used to determine if the agent
is a callback agent or not. (closes issue #14173) Reported by:
nathan Patches: 14173.patch uploaded by putnopvut (license 60)
Tested by: nathan, aramirez ........
* /, main/slinfactory.c: Merged revisions 171621 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r171621 | mmichelson | 2009-01-27 14:06:01 -0600 (Tue, 27 Jan
2009) | 18 lines Prevent a crash from occurring when a jitter
buffer interpolated frame is removed from a slinfactory
slinfactory used the "samples" field of an ast_frame in order to
determine the amount of data contained within the frame. In
certain cases, such as jitter buffer interpolated frames, the
frame would have a non-zero value for "samples" but have NULL
"data" This caused a problem when a memcpy call in
ast_slinfactory_read would attempt to access invalid memory. The
solution in use here is to never feed frames into the slinfactory
if they have NULL "data" (closes issue #13116) Reported by:
aragon Patches: 13116.diff uploaded by putnopvut (license 60)
........
* apps/app_queue.c: Fix queue crashes that would occur after the
calling channel was masqueraded. The data passed to the
end_bridge_callback was assumed to be data which was still
stack'd. The problem was that with some call features, attended
transfers in particular, a new bridge thread is started once the
feature completes, meaning that when the end_bridge_callback is
called, the end_bridge_callback_data was invalid. To fix this
problem, there are two measures taken 1. Instead of pointing to
stacked data, we now used heap-allocated data for passing to the
end_bridge_callback in app_queue 2. Since bridges can end
multiple times on a single logical call, we wait until the final
bridge is broken to actually set any queue variables. This is
accomplished through reference-counting and the use of an
end_bridge_callback_data_fixup function in app_queue.c (closes
issue #14260) Reported by: ccesario Patches: 14260.patch uploaded
by putnopvut (license 60) Tested by: ccesario
2009-01-27 15:23 +0000 [r171558] Doug Bailey <dbailey@digium.com>
* channels/chan_dahdi.c: Handle new VMWI ioctl structure (Now there
are two VMWI ioctl calls.) (issue #14104) Reported by: alecdavis
Tested by: dbailey
2009-01-27 15:00 +0000 [r171263-171528] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Solving the same issue, but a bit
different in trunk... Merged revisions 171527 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r171527 | oej | 2009-01-27 15:33:20 +0100 (Tis, 27 Jan 2009) | 13
lines Use the same branch tag in CANCEL as in INVITE Originally
putnopvut implemented some changes in revision 142079 that
according to the bug report seemed to have worked then, but
somehow fails now. I guess code, as humans, get old and forget
stuff. Anyway, this bug caused CANCEL not to work with picky
systems. Thanks Fredrik for pointing out where the bug in the SIP
messaging was. (closes issue #14346) Reported by: oej Patches:
bug14346.diff uploaded by oej (license 306) Tested by: oej
........
* channels/chan_sip.c: Moving generic setting to friends
* channels/chan_sip.c: Continue to move variables into the sip_cfg
structure to make them easier to handle in the future as a group
of settings for a group of devices. At some point, I want one
sip_cfg per domain handled, so we can have "group" settings.
* channels/chan_sip.c: Just moving around variable declarations so
that we have all globals in the same place. Default setting is
set before we activate the channel or at reloads, not where we
declare the variable.
* /, channels/chan_sip.c: Merged revisions 171264 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r171264 | oej | 2009-01-26 13:51:53 +0100 (MÃ¥n, 26 Jan 2009) | 9
lines Don't retransmit 401 on REGISTER requests when
alwaysauthreject=yes (closes issue #14284) Reported by: klaus3000
Patches: patch_chan_sip_unreliable_1.4.23_14284.txt uploaded by
klaus3000 (license 65) Tested by: klaus3000 ........
* main/channel.c: Add extensions and context on manager event when
new channel is created.
2009-01-25 23:58 +0000 [r171188] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_oss.c: Merged revisions 171187 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r171187 | tilghman | 2009-01-25 17:44:01 -0600 (Sun, 25 Jan 2009)
| 6 lines Correctly track the hookstate (closes issue #13686)
Reported by: itiliti Patches: 20081013__bug13686.diff.txt
uploaded by Corydon76 (license 14) ........
2009-01-25 16:50 +0000 [r171043-171081] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_skinny.c: dont segfault when a MWI event occurs on
a line without a registered device
* configs/skinny.conf.sample: Make the sample skinny.conf work
(closes issue #14325) Reported by: DEA Patches:
skinny.conf.sample-trunk.txt uploaded by DEA (license 3)
2009-01-25 13:35 +0000 [r170980] Sean Bright <sean.bright@gmail.com>
* /, apps/app_page.c: Merged revisions 170979 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r170979 | seanbright | 2009-01-25 08:33:20 -0500 (Sun, 25 Jan
2009) | 9 lines Resolve a logic error that was causing Page() to
crash when more than one channel was specified. (closes issue
#14308) Reported by: bluefox Patches: 20090124__bug14308.diff.txt
uploaded by seanbright (license 71) Tested by: kc0bvu ........
2009-01-25 02:49 +0000 [r170902-170943] Russell Bryant <russell@digium.com>
* include/asterisk/utils.h: Change ARRAY_LEN() to be more C++ safe.
When the second part of this macro is written as 0[a] instead of
a[0], it will force a failure if the macro is used on a C++
object that overloads the [] operator.
* res/res_agi.c: Add a todo to finish the XML docs in this module
2009-01-24 13:55 +0000 [r170837] Tilghman Lesher <tlesher@digium.com>
* /, configs/res_odbc.conf.sample: Merged revisions 170836 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r170836 | tilghman | 2009-01-24 07:55:02 -0600 (Sat, 24 Jan 2009)
| 2 lines Remove superfluous implementation note (closes issue
#14319) ........
2009-01-23 23:10 +0000 [r170794] Richard Mudgett <rmudgett@digium.com>
* doc/tex/Makefile: Fix asterisk.pdf generation if branch name has
an underscore in it.
2009-01-23 22:58 +0000 [r170790] Russell Bryant <russell@digium.com>
* doc/tex/Makefile: Don't blow up if a branch name has an
underscore in it
2009-01-23 20:56 +0000 [r170677-170720] Mark Michelson <mmichelson@digium.com>
* /, configs/res_odbc.conf.sample: Merged revisions 170719 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r170719 | mmichelson | 2009-01-23 14:55:26 -0600 (Fri, 23 Jan
2009) | 8 lines Add notes to the idlecheck explanation in
res_odbc.conf.sample (closes issue #14319) Reported by: klaus3000
Patches: patch_idlecheck_res_odbc.conf.sample.txt uploaded by
klaus3000 (license 65) ........
* /, contrib/i18n.testsuite.conf: Merged revisions 170671 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r170671 | mmichelson | 2009-01-23 14:21:51 -0600 (Fri, 23 Jan
2009) | 14 lines Update contrib/i18n.testsuite.conf to not use
deprecated syntax * Convert Wait,1 to Wait(1) * Convert
SetLanguage to Set(CHANNEL(language)) * Use 'n' for all
priorities beyond the first Also added test for Chinese numbers,
too. (closes issue #14320) Reported by: dant Patches:
i18n.testsuite.conf.issue14320.v2.diff uploaded by dant (license
670) ........
2009-01-23 20:18 +0000 [r170652] Joshua Colp <jcolp@digium.com>
* main/channel.c, /: Merged revisions 170648 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r170648 | file | 2009-01-23 16:16:39 -0400 (Fri, 23 Jan 2009) | 4
lines When a channel is answered make sure any indications
currently playing stop. Usually the phone would do this but if
the channel was already answered then they are being generated by
Asterisk and we darn well need to stop them. (closes issue
#14249) Reported by: RadicAlish ........
2009-01-23 19:25 +0000 [r170608] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_iax2.c: Merged revisions 170588 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r170588 | tilghman | 2009-01-23 13:20:44 -0600 (Fri, 23
Jan 2009) | 2 lines Additions to AST-2009-001 ........
2009-01-23 19:09 +0000 [r170505-170569] Joshua Colp <jcolp@digium.com>
* apps/app_dial.c, /: Merged revisions 170568 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r170568 | file | 2009-01-23 15:06:54 -0400 (Fri, 23 Jan 2009) | 4
lines When a call is forwarded stop any active indications. The
new channel will provide an indication, if need be, itself.
(closes issue #14310) Reported by: RadicAlish ........
* /, channels/chan_sip.c: Merged revisions 170504 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r170504 | file | 2009-01-23 14:04:08 -0400 (Fri, 23 Jan 2009) | 4
lines Use the on hold flag to see if the call is on hold or not.
It is possible that our address for them will still be valid even
though they are on hold. (closes issue #14295) Reported by:
klaus3000 ........
2009-01-23 17:46 +0000 [r170501] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_h323.c: let's use SENTINEL where needed
2009-01-23 17:32 +0000 [r170498] Joshua Colp <jcolp@digium.com>
* apps/app_voicemail.c: Reset the ast_str used for escape
substitution. We need to do this since it is a thread local
variable that may contain the value of a previous substitution.
(closes issue #14312) Reported by: pj
2009-01-23 17:03 +0000 [r170463] Matthew Fredrickson <creslin@digium.com>
* channels/chan_dahdi.c: We should not do restart messages if we're
in PTMP mode
2009-01-23 16:57 +0000 [r170460] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_skinny.c: Dont clear the display of skinny phones
when not needed. (closes issue #13182) Reported by: pj Patches:
2009011901_dontcleardisplay.diff.txt uploaded by mvanbaak
(license 7) Tested by: mvanbaak, pj
2009-01-23 16:35 +0000 [r170457] Doug Bailey <dbailey@digium.com>
* channels/chan_dahdi.c: MWI messages included in CID spill was not
being properly handled and prevented the call from being
processed (issue #14313) Reported by: seandarcy Tested by:
dbailey
2009-01-23 15:44 +0000 [r170393] Mark Michelson <mmichelson@digium.com>
* main/channel.c, /: Merged revisions 170392 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r170392 | mmichelson | 2009-01-23 09:40:39 -0600 (Fri, 23 Jan
2009) | 28 lines Fix broken call pickup There was a subtle change
in ast_do_masquerade which resulted in failed attempts to pickup
calls. The problem was that the value of the AST_FLAG_OUTGOING
flag was copied from the clone to the original channel. In the
case of call pickup, this meant that the AST_FLAG_OUTGOING flag
ended up being cleared on the channel that was attempting to
execute the pickup. Because this flag was not set, when ast_read
came across an answer frame, it ignored it. The result of this
was that the calling channel was never properly answered. This
fix changes the behavior in ast_do_masquerade to set the flags on
the original channel to the union of the flags on the clone
channel. This way, if the AST_FLAG_OUTGOING flag is set on either
of the two channels involved in the masquerade, the resulting
channel will have the flag set as well. (closes issue #14206)
Reported by: francesco_r Patches: 14206.patch uploaded by
putnopvut (license 60) Tested by: francesco_r, aragon, putnopvut
........
2009-01-22 23:23 +0000 [r170351] Matthew Fredrickson <creslin@digium.com>
* channels/chan_dahdi.c: Make sure we don't set the channel to be
inalarm for a D-channel drop on PTMP connections
2009-01-22 21:25 +0000 [r170307] Tilghman Lesher <tlesher@digium.com>
* main/abstract_jb.c: Create logfile safely. (closes issue #14160)
Reported by: tzafrir Patches: 20090104__bug14160.diff.txt
uploaded by Corydon76 (license 14)
2009-01-22 20:04 +0000 [r170240] Joshua Colp <jcolp@digium.com>
* /, main/rtp.c: Merged revisions 170239 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r170239 | file | 2009-01-22 16:02:35 -0400 (Thu, 22 Jan 2009) | 7
lines Don't crash if RTCP is not enabled on an RTP structure but
statistics are output. (closes issue #14234) Reported by: jcovert
Patches: rtp.c.patch-1.6.0.3 uploaded by jcovert (license 551)
rtp.c.patch-svn-165599 uploaded by jcovert (license 551) ........
2009-01-22 17:19 +0000 [r170165] Tilghman Lesher <tlesher@digium.com>
* /, pbx/pbx_config.c: Merged revisions 170158 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r170158 | tilghman | 2009-01-22 11:18:07 -0600 (Thu, 22 Jan 2009)
| 6 lines Allow global variables after substitution to be as long
as other variables. (closes issue #14263) Reported by: markd
Patches: 20090120__bug14263.diff.txt uploaded by Corydon76
(license 14) ........
2009-01-22 16:52 +0000 [r170148] Joshua Colp <jcolp@digium.com>
* /, apps/app_meetme.c: Merged revisions 170147 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r170147 | file | 2009-01-22 12:50:54 -0400 (Thu, 22 Jan 2009) | 4
lines If we are unable to request a DAHDI pseudo channel and we
are using the user introduction without review option make sure
it gets unset so other code does not blindly assume a DAHDI
pseudo channel exists. (closes issue #14282) Reported by:
cheesegrits ........
2009-01-22 15:49 +0000 [r170112] Doug Bailey <dbailey@digium.com>
* channels/chan_dahdi.c, configure,
include/asterisk/autoconfig.h.in, configure.ac: change VMWI to
use new DAHDI_VMWI ioctl call. Change configure script to detect
the new ioctl call data structure. (issue #14104) Reported by:
alecdavis Patches: mwiioctl_structure_asterisk.diff4.txt uploaded
by dbailey (license ) Tested by: alecdavis, dbailey
2009-01-22 15:14 +0000 [r170047-170051] Joshua Colp <jcolp@digium.com>
* main/pbx.c, /: Merged revisions 170050 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r170050 | file | 2009-01-22 11:13:56 -0400 (Thu, 22 Jan 2009) | 6
lines Do a string comparison instead of pointer comparison since
some people specify the context they are actually in as an
argument to get around some funkiness. (closes issue #14011)
Reported by: dveiga Patches: pbx.c.patch uploaded by dveiga
(license 665) ........
* apps/app_parkandannounce.c: Clear the autoloop flag when parsing
and setting the context/extension/priority to go back to. When
the channel executes a PBX again we want it to start out at the
point we explicitly say and at that point it will not yet be
doing autoloop. (closes issue #14304) Reported by: jcovert
2009-01-22 02:10 +0000 [r170007] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: * Adjust some conditionals to balance
curly braces. * Other minor changes.
2009-01-22 00:44 +0000 [r169944] Tilghman Lesher <tlesher@digium.com>
* /, include/asterisk/linkedlists.h: Merged revisions 169943 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r169943 | tilghman | 2009-01-21 18:43:31 -0600 (Wed, 21 Jan 2009)
| 9 lines AST_RWLOCK_INIT_VALUE is always defined. What we really
wanted to ask is whether autoconf detected a static initializer
value. This fixes rwlocks on all such platforms (mainly, Mac OS
X). (closes issue #13767) Reported by: jcovert Patches:
20090121__bug13767.diff.txt uploaded by Corydon76 (license 14)
Tested by: jcovert, Corydon76 ........
2009-01-22 00:23 +0000 [r169910] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: Whitespace changes only
2009-01-21 23:25 +0000 [r169869] Joshua Colp <jcolp@digium.com>
* main/pbx.c, /: Merged revisions 169867 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r169867 | file | 2009-01-21 19:20:47 -0400 (Wed, 21 Jan 2009) | 4
lines Read lock the contexts to maintain the locking order when
we are notified that the state of a device has changed. (closes
issue #13839) Reported by: mcallist ........
2009-01-21 23:20 +0000 [r169794-169866] Mark Michelson <mmichelson@digium.com>
* channels/chan_dahdi.c: Test commit for test issue #14303
* main/say.c: Fix a crash when saying certain numbers in Chinese
This commit fixes a crash that was occurring when attempting to
say a number between 10000 and 100000 due to dividing by 0. This
also removes some places where a "zero" is spoken when it should
not be. (closes issue #14291) Reported by: dant Patches:
say.c-14291.diff uploaded by dant (license 670) Tested by: dant
2009-01-21 22:04 +0000 [r169793] Michiel van Baak <michiel@vanbaak.info>
* doc/tex/extensions.tex: remove duplicated sentence.
2009-01-21 21:53 +0000 [r169791] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Further fix some oddities in sip show users
and sip show peers logic ccesario on IRC pointed out that his sip
peers were not displayed properly when he would issue the command
"sip show peers." The problem was that the onlymatchonip field
was used to determine if the endpoint was a "peer" or "user." The
tricky part is that a "friend" is supposed to be treated as both
a "user" and a "peer" but the logic would not allow "friends" to
show up as "peers" since onlymatchonip was set to FALSE for
friends. I have modified the sip_peer structure to more
explicitly keep track of what type endpoint it is so that the
various manager and CLI commands will display the expected
information Reported by ccesario via IRC Tested by ccesario
2009-01-21 21:03 +0000 [r169723] Tilghman Lesher <tlesher@digium.com>
* /, main/asterisk.c: Merged revisions 169722 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r169722 | tilghman | 2009-01-21 15:02:32 -0600 (Wed, 21 Jan 2009)
| 8 lines Extra NULLs in the output cause some terminal types to
abort in the middle of a color code, causing terminal weirdness.
(closes issue #14130) Reported by: coolmig Patches:
20090121__bug14130.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76, coolmig ........
2009-01-21 17:21 +0000 [r169673] Steve Murphy <murf@digium.com>
* utils/refcounter.c: This patch corrects a segfault reported in
14289, due to a null ptr being refd. Yes, seanbright is right in
the bug comments, that is the fix. Sorry for this oversight; I
guess my personal usage didn't have this happen! murf (closes
issue #14289) Reported by: jamesgolovich
2009-01-21 10:49 +0000 [r169620-169625] Russell Bryant <russell@digium.com>
* /: Remove properties that erroneously got merged into trunk
* main/tcptls.c: Fix a regression in TCP support. This patch fixes
a problem that caused chan_sip to think that every open TCP
session was to a remote address of 0.0.0.0:0. (closes issue
#14287) Reported by: jamesgolovich Patches: bug-14287.diff.txt
uploaded by jamesgolovich (license 176)
2009-01-21 00:33 +0000 [r169557-169611] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Fix device state parsing issues for channel
names with multiple slashes The fix being applied is a bit
different for trunk and the 1.6.X branches. For trunk, we only
wish to strip off the characters beyond the second slash if the
channel is a Local channel (i.e. we are removing the /n from the
device name). Other channel technologies with multiple slashes
(e.g. DAHDI) need the information after the second slash in order
to get the proper device state information. In addition to this
fix, the 1.6.X branches are receiving a much more important fix
as well. The problem in 1.6.X is that the member's device name
was being directly changed instead of having a copy changed. This
meant that we would strip off the second slash and trailing
characters and then leave the member's device name like that
permanently thereafter. (closes issue #14014) Reported by:
kebl0155 Patches: 14014_number2.patch uploaded by putnopvut
(license 60) Tested by: kebl0155
* apps/app_queue.c: Use the default timeout for a queue instead of
-1 (closes issue #14272) Reported by: timking
* /, channels/chan_sip.c: Convert the character pointers in a
sip_request to be pointer offsets When an ast_str expands to hold
more data, any pointers that were pointing to the data prior to
the expansion will be pointing at invalid memory. This change
makes such pointers used in chan_sip.c instead be offsets from
the beginning of the string so that the same math may be applied
no matter where in memory the string resides. To help ease this
transition, a macro called REQ_OFFSET_TO_STR has been added to
chan_sip.c so that given a sip_request and an offset, the string
at that offset is returned. (closes issue #14220) Reported by:
riksta Tested by: putnopvut Review
http://reviewboard.digium.com/r/126/
2009-01-20 19:22 +0000 [r169486-169510] Terry Wilson <twilson@digium.com>
* main/features.c: Make a proper builtin attended transfer to
parking work This is an ugly hack from 1.4 that allows the
timeout callback from a parked call to use the right channel name
for the callback when the park is done with a builtin attended
transfer (that isn't completed early). This hasn't ever worked in
trunk and no one has complained yet, so eh.
* /, main/features.c: Merged revisions 169485 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r169485 | twilson | 2009-01-20 12:40:56 -0600 (Tue, 20 Jan 2009)
| 6 lines Don't play audio to the channel if we've masqueraded
(closes issue #14066) Reported by: bluefox Tested by:
otherwiseguy, bluefox ........
2009-01-19 21:42 +0000 [r169438] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk/res_odbc.h, funcs/func_odbc.c,
include/asterisk/strings.h, res/res_odbc.c: ast_str_SQLGetData is
*not* part of the ast_str API, it's part of the ast_odbc API and
just happens to use an ast_str as the buffer; move all of it to
res_odbc.c and res_odbc.h, renaming appropriately along the way
fix some minor coding style issues in strings.h and add some
attribute_pure annotations to functions in the ast_str API
2009-01-19 20:14 +0000 [r169367-169369] Michiel van Baak <michiel@vanbaak.info>
* main/asterisk.c: fix assignment in swapmode plug. Spotted and fix
provided by ys (closes issue #14129) Reported by: ys Tested by:
ys
* channels/chan_skinny.c: Redo the event-based MWI in chan_skinny.
Dan saw regular segfaults with the old implementation and rewrote
it to make it really eventbased. I altered it to be trunk
compatible and wedhorn gave some feedback and ideas how to make
it even better. (closes issue #13821) Reported by: DEA Patches:
chan_skinny-mwi-events.txt uploaded by DEA (license 3) Tested by:
mvanbaak, DEA "no probs by me" from wedhorn
2009-01-19 20:05 +0000 [r169365] Tilghman Lesher <tlesher@digium.com>
* main/manager.c, /, apps/app_userevent.c: Merged revisions 169364
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r169364 | tilghman | 2009-01-19 13:49:25 -0600 (Mon, 19 Jan 2009)
| 4 lines Truncate userevents at the end of a line, when the
command exceeds the buffer. (closes issue #14278) Reported by:
fnordian ........
2009-01-19 18:36 +0000 [r169327] Michiel van Baak <michiel@vanbaak.info>
* main/asterisk.c: Make asterisk compile on non-amd64 versions of
OpenBSD. The HW_PHYSMEM64 is only available in latest OpenBSD
and/or amd64 versions of OpenBSD. Use HW_PHYSMEM when
HW_PHYSMEM64 is not available. (closes issue #14129) Reported by:
ys Patches: 2009011600_physmem64.diff.txt uploaded by mvanbaak
(license 7) Tested by: mvanbaak, jtodd
2009-01-19 18:22 +0000 [r169277-169325] Doug Bailey <dbailey@digium.com>
* channels/chan_dahdi.c: Get rid of magic number and replace with
DAHDI_VMWI_NUMBER_MASK when determining the number of messages
pending for MWI call
* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Add
enhanced MWI generation to take advantage of new dahdi line
reversal MWI ability. (closes issue #14104) Reported by:
alecdavis Patches: asttrunk-14104.diff2.txt uploaded by dbailey
(license ) chan_dahdi.rpas_and_fsk.diff.txt uploaded by alecdavis
(license 585) Tested by: alecdavis, dbailey
2009-01-19 15:54 +0000 [r169211] Mark Michelson <mmichelson@digium.com>
* channels/chan_local.c, /: Merged revisions 169210 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r169210 | mmichelson | 2009-01-19 09:52:15 -0600 (Mon,
19 Jan 2009) | 13 lines Prevent a crash in chan_local due to a
potential NULL pointer dereference Move the check for if both
channels on a local_pvt have generators to below where p->chan is
checked for NULLity (NULLness?). This prevents a crash from
occurring if p->chan is NULL. (closes issue #14189) Reported by:
sascha Patches: 14189.patch uploaded by putnopvut (license 60)
Tested by: sascha ........
2009-01-17 18:26 +0000 [r169153] Doug Bailey <dbailey@digium.com>
* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Add
discriminator for when ring pulse alert signal is used to preface
MWI spills This prevents the situation when MWI messages are
added to caller ID spills causing the channel to be hung up
2009-01-17 02:52 +0000 [r169116] Sean Bright <sean.bright@gmail.com>
* pbx/pbx_dundi.c: Change intializer types. Found while working on
asterisk-cpp. I have a new favorite error message from g++:
pbx_dundi.c:4580: sorry, unimplemented: non-trivial designated
initializers not supported I like it when compilers are
apologetic.
2009-01-17 01:56 +0000 [r169044-169080] Terry Wilson <twilson@digium.com>
* main/tcptls.c, main/http.c, include/asterisk/tcptls.h: Fix
qualify for TCP peer (closes issue #14192) Reported by:
pabelanger Patches: asterisk-bug14192.diff.txt uploaded by
jamesgolovich (license 176) Tested by: jamesgolovich
* channels/chan_sip.c: Fix port :0 added to SIP INVITE URI when
outboundproxy used (closes issue #14233) Reported by: chris-mac
Patches: asterisk-bug14233.diff.txt uploaded by jamesgolovich
(license 176) Tested by: jamesgolovich, chris-mac, otherwiseguy
2009-01-16 22:43 +0000 [r168976] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 168975 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r168975 | mmichelson | 2009-01-16 16:42:13 -0600 (Fri, 16 Jan
2009) | 18 lines Account for possible NULL pointer when we
receive a 408 in response to a REGISTER It may be that by the
time we receive a reply to a REGISTER request, the attempt has
timed out and thus the registry structure pointed to by the
corresponding sip_pvt has gone away. This situation was handled
properly for a 200 OK response, but the 408 case assumed that the
sip_registry struct was non-NULL, thus potentially causing a
crash This commit fixes this assumption and prints out a message
to the console if we should receive a late 408 response to a
REGISTER (closes issue #14211) Reported by: aborghi Patches:
14211.diff uploaded by putnopvut (license 60) Tested by: aborghi
........
2009-01-16 22:16 +0000 [r168941] Terry Wilson <twilson@digium.com>
* /, main/features.c: Merged revisions 168716 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r168716 | twilson | 2009-01-15 12:22:49 -0600 (Thu, 15 Jan 2009)
| 12 lines Convert call to park_call_full to
masq_park_call_announce Since we removed the AST_PBX_KEEPALIVE
return value, we need to use masqueraded parking, otherwise we
will try to call ast_hangup() in __pbx_run() and in
do_parking_thread() and then promptly crash. (closes issue
#14215) Reported by: waverly360 Tested by: otherwiseguy (closes
issue #14228) Reported by: kobaz Tested by: otherwiseguy ........
2009-01-16 19:54 +0000 [r168898] Mark Michelson <mmichelson@digium.com>
* res/res_timing_timerfd.c: Fix a logic error that occur when using
the timerfd interface This sequence of events posed a problem
timerfd_timer_open timerfd_timer_enable_continuous
timerfd_timer_set_rate timerfd_timer_disable_continuous The
reason was that the timing module was written under the
assumption that timerfd_timer_set_rate would not be called
between enabling and disabling continuous mode. What happened in
this situation was that timerfd_timer_enable_continuous saved off
our previously set timer (in this situation a 0 timer, meaning it
never runs out). Then timerfd_timer_disable_continuous would
restore this 0 timer, even though it logically should set the
timer to be whatever was set in timerfd_timer_set_rate. Now the
behavior in timerfd_timer_set_rate is to overwrite the saved
timer that may or may not have been set in
timerfd_timer_enable_continuous. Even if
timerfd_timer_enable_continuous has not been previously called,
this will not harm the operation. Thanks to Terry Wilson for
discovering the problem and giving me a really great debug
capture that pointed out the problem clearly
2009-01-16 18:49 +0000 [r168832] Tilghman Lesher <tlesher@digium.com>
* /, main/say.c, include/asterisk/say.h, apps/app_voicemail.c:
Merged revisions 168828 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r168828 | tilghman | 2009-01-16 12:41:35 -0600 (Fri, 16 Jan 2009)
| 6 lines Fix the conjugation of Russian and Ukrainian languages.
(related to issue #12475) Reported by: chappell Patches:
vm_multilang.patch uploaded by chappell (license 8) ........
2009-01-16 17:09 +0000 [r168759-168760] Russell Bryant <russell@digium.com>
* CHANGES: Fix a spelling mistake.
* channels/chan_misdn.c: build in dev mode
2009-01-16 00:34 +0000 [r168737-168746] Steve Murphy <murf@digium.com>
* res/ael/pval.c, /: Merged revisions 168745 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r168745 | murf | 2009-01-15 17:19:12 -0700 (Thu, 15 Jan 2009) |
14 lines This patch fixes a problem where a goto (or jump, in
this case) fails a consistency check because it can't find a
matching extension. The problem was a missing instruction to end
the range notation in the code where it converts the pattern into
a regex and uses the regex code to determine the match. I tested
using the AEL code the user supplied, and now, the consistency
check passes. (closes issue #14141) Reported by: dimas ........
* main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2.c: This patch
allows null args in ast_expr2 func calls, and fixes commas being
converted to pipes, which was 1.4 type stuff. If the user says
count=ENUMLOOKUP(${EXTEN},ALL,c,,enum.mydomain.tld); then it
won't complain about the empty arg (c,,...) and fabled's patch
won't let it swap the commas for pipes. Ran it thru my dialplan
and no complaints. (closes issue #14169) Reported by: fabled
Patches: function-argument-separator-fix.diff uploaded by fabled
(license 448)
2009-01-15 20:18 +0000 [r168734] Kevin P. Fleming <kpfleming@digium.com>
* res/res_config_odbc.c, build_tools/menuselect-deps.in, configure,
funcs/func_odbc.c, configure.ac, cdr/cdr_adaptive_odbc.c,
cdr/cdr_odbc.c, makeopts.in, res/res_odbc.c,
apps/app_voicemail.c: remove the PBX_ODBC logic from the
configure script, and add GENERIC_ODCB logic that includes
copying the relevant LIB and INCLUDE data from either UnixODBC or
iODBC, based on which was found; if both were found, prefer
UnixODBC this stops modules from being linked against both sets
of libraries on systems that have both installed
2009-01-15 20:00 +0000 [r168725-168732] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Add missing brace
* channels/chan_sip.c: Fix the compactheaders option in sip.conf
* channels/chan_sip.c: Remove an unneeded condition for line
addition to a SIP request/response In Asterisk 1.4 and 1.6.0, the
sip_request structure had a statically allocated buffer to hold
the text of the request. There was a check in the add_line
function to not attempt to write the line into the buffer if we
did not have room for it. In trunk and Asterisk versions starting
with 1.6.1, an expandable ast_str structure is used to hold the
text. Since it may grow to fit an arbitrarily sized string, this
check in add_line is no longer valid. I found this oddity while
attempting to fix issue #14220; however, I do not believe that
this is the fix for that issue since the output supplied by the
reporter did not contain the warning message that would be
printed had this condition been satisfied.
2009-01-15 18:47 +0000 [r168722] Olle Johansson <oej@edvina.net>
* /, configs/extconfig.conf.sample: Merged revisions 168721 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r168721 | oej | 2009-01-15 19:43:43 +0100 (Tor, 15 Jan 2009) | 2
lines Meetme actually has realtime but wasn't documented ........
2009-01-15 18:39 +0000 [r168719] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/strings.h: Resolve issue with negative vs
non-negative length parameters. (closes issue #14245) Reported
by: dveiga
2009-01-15 18:08 +0000 [r168711-168712] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Make sure that we have the same terminology
in sip.conf.sample and the source code warning. Thanks Nick Lewis
for pointing this out in the bug tracker.
* configs/sip.conf.sample: Clarify some misunderstandings and make
it even more clear that you can refer to a peer in the register=
line.
2009-01-15 15:33 +0000 [r168705] Sean Bright <sean.bright@gmail.com>
* apps/app_meetme.c: Add a missing unlock and properly handle the
'maxusers' setting on MeetMe conferences. We were using the 'user
number' field to compare against the maximum allowed users, which
works assuming users with lower user numbers didn't leave the
conference. (closes issue #14117) Reported by: sergedevorop
Patches: 20090114__bug14117-2.diff.txt uploaded by seanbright
(license 71) Tested by: sergedevorop
2009-01-15 13:37 +0000 [r168636-168639] Olle Johansson <oej@edvina.net>
* CREDITS, CHANGES: Related to issue #14246 Update changes for
SIPRemoveHeader()
* channels/chan_sip.c: Add capability to remove added SIP headers
*before* INVITE is generated. (closes issue #14246) Reported by:
klaus3000 Patches: 2patch_chan_sip_SIPRemoveHeader_trunk.txt
uploaded by klaus3000 (license 65)
* apps/app_queue.c: Add support for setting the Reason header when
cancelling a call in the queue because someone else answered.
Previously, only dial() was supported. EDV-102
2009-01-15 00:14 +0000 [r168629] Mark Michelson <mmichelson@digium.com>
* /, apps/app_queue.c: Merged revisions 168628 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r168628 | mmichelson | 2009-01-14 18:11:01 -0600 (Wed, 14 Jan
2009) | 16 lines Fix some crashes from bad datastore handling in
app_queue.c * The queue_transfer_fixup function was searching for
and removing the datastore from the incorrect channel, so this
was fixed. * Most datastore operations regarding the
queue_transfer datastore were being done without the channel
locked, so proper channel locking was added, too. (closes issue
#14086) Reported by: ZX81 Patches: 14086v2.patch uploaded by
putnopvut (license 60) Tested by: ZX81, festr ........
2009-01-14 23:10 +0000 [r168626] Sean Bright <sean.bright@gmail.com>
* main/cli.c: Don't crash when typing 'core set verbose' or 'core
set debug' by themselves. (closes issue #14219) Reported by:
jamesgolovich Patches: asterisk-setverbosecrash.diff.txt uploaded
by jamesgolovich (license 176)
2009-01-14 21:51 +0000 [r168623] Richard Mudgett <rmudgett@digium.com>
* /, channels/misdn/isdn_lib.c: Merged revisions 168622 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r168622 | rmudgett | 2009-01-14 15:48:22 -0600 (Wed, 14 Jan 2009)
| 4 lines * Fixed create_process() allocation of process ID
values. The allocated process IDs could overflow their respective
NT and TE fields. Affects outgoing calls. ........
2009-01-14 21:19 +0000 [r168619] Doug Bailey <dbailey@digium.com>
* channels/chan_dahdi.c: This fixes a problem where MWI FSK spills
were being injected onto off hook fxs lines. (closes issue
#14143) Reported by: alecdavis Patches:
chan_dahdi-14143.patch.txt uploaded by dbailey (license ) Tested
by: alecdavis
2009-01-14 20:58 +0000 [r168615] Sean Bright <sean.bright@gmail.com>
* /, contrib/scripts/autosupport: Merged revisions 168614 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r168614 | seanbright | 2009-01-14 15:52:00 -0500 (Wed, 14 Jan
2009) | 9 lines Update autosupport script to supply info for both
Zaptel and DAHDI in 1.4 and be sure to run dahdi_test in 1.6.x
and trunk instead of zttest. (closes issue #14132) Reported by:
dsedivec Patches: asterisk-1.4-autosupport.patch uploaded by
dsedivec (license 638) asterisk-trunk-autosupport.patch uploaded
by dsedivec (license 638) ........
2009-01-14 20:51 +0000 [r168613] Steve Murphy <murf@digium.com>
* /, apps/app_page.c: Merged revisions 168608 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r168608 | murf | 2009-01-14 12:34:35 -0700 (Wed, 14 Jan 2009) | 1
line app_page was failing to compile in dev-mode on my gcc-4.2.4
system. This change gets rid of the warning. ........
2009-01-14 20:13 +0000 [r168610] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Restore the "sip show users" and "sip show
user" CLI commands (closes issue #14180) Reported by: amorsen
Patches: sip_show_users_161v3.diff uploaded by putnopvut (license
60) Tested by: blitzrage, amorsen
2009-01-14 19:36 +0000 [r168609] Michiel van Baak <michiel@vanbaak.info>
* main/asterisk.c: Fix compilation on FreeBSD and OSX This started
as work to fix the 'core show sysinfo' CLI command but while
working on it oej pointed out that read_credentials did not
compile neither. So while being there, fix that as well. Thanks
for all the testing oej! (closes issue #14129) Reported by: ys
Tested by: oej, mvanbaak
2009-01-14 19:11 +0000 [r168601-168604] Tilghman Lesher <tlesher@digium.com>
* main/udptl.c, /: Merged revisions 168603 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r168603 | tilghman | 2009-01-14 13:02:55 -0600 (Wed, 14 Jan 2009)
| 7 lines Don't read into a buffer without first checking if a
value is beyond the end. (closes issue #13600) Reported by: atis
Patches: 20090106__bug13600.diff.txt uploaded by Corydon76
(license 14) Tested by: atis ........
* channels/chan_misdn.c: Mostly spacing changes; no functionality
change at all.
2009-01-14 02:00 +0000 [r168594] Terry Wilson <twilson@digium.com>
* /, apps/app_page.c: Merged revisions 168593 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r168593 | twilson | 2009-01-13 19:27:18 -0600 (Tue, 13 Jan 2009)
| 20 lines Don't overflow when paging more than 128 extensions
The number of available slots for calls in app_page was hardcoded
to 128. Proper bounds checking was not in place to enforce this
limit, so if more than 128 extensions were passed to the Page()
app, Asterisk would crash. This patch instead dynamically
allocates memory for the ast_dial structures and removes the
(non-functional) arbitrary limit. This issue would have special
importance to anyone who is dynamically creating the argument
passed to the Page application and allowing more than 128
extensions to be added by an outside user via some external
interface. The patch posted by a_villacis was slightly modified
for some coding guidelines and other cleanups. Thanks,
a_villacis! (closes issue #14217) Reported by: a_villacis
Patches: 20080912-asterisk-app_page-fix-buffer-overflow.patch
uploaded by a (license 660) Tested by: otherwiseguy ........
2009-01-13 23:57 +0000 [r168591] Tilghman Lesher <tlesher@digium.com>
* channels/chan_misdn.c: Janitor patch for chan_misdn (make channel
variable access safe) (closes issue #12887) Reported by: pputman
Patches: chan_misdn_threadsafe.patch uploaded by pputman (license
81)
2009-01-13 23:05 +0000 [r168585-168588] Terry Wilson <twilson@digium.com>
* res/res_http_post.c: Fully overwrite a same-named file when
uploading (closes issue #14190) Reported by: timking
* Makefile, include/asterisk/options.h, main/asterisk.c: Add option
to hide console connect messages (closes issue #14222) Reported
by: jamesgolovich Patches: asterisk-hideconnect.diff.txt uploaded
by jamesgolovich (license 176) Tested by: otherwiseguy
2009-01-13 22:30 +0000 [r168579] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Clarify a message that app_queue prints and
change to a debug-level message The "No one is answering..."
verbose message contained 3 numbers that were not explained in
any way to whoever was viewing the message. It is more helpful
now since the message explains what the numbers mean. Also, the
message has been downgraded to "DEBUG" level. (closes issue
#14172) Reported by: caio1982 Patches: queue_answering_debug.diff
uploaded by caio1982 (license 22)
2009-01-13 22:22 +0000 [r168578] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c: Merged revisions 168551 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r168551 | twilson | 2009-01-13 12:34:14 -0600 (Tue, 13 Jan 2009)
| 7 lines Don't pass a value with a side effect to a macro
(closes issue #14176) Reported by: paraeco Patches:
chan_sip.c.diff uploaded by paraeco (license 658) ........
2009-01-13 21:18 +0000 [r168575] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Allow
specifying a port number in the user portion of a register =>
line in sip.conf With this commit, a register => line in sip.conf
may contain a port number in the "user" section of the line.
Please see CHANGES and sip.conf.sample for more details regarding
this. (closes issue #14198) Reported by: Nick_Lewis Patches:
chan_sip.c-domainport2.patch uploaded by Nick (license 657)
Tested by: Nick_Lewis
2009-01-13 19:22 +0000 [r168562] Russell Bryant <russell@digium.com>
* channels/chan_unistim.c, main/pbx.c, apps/app_read.c, /,
include/asterisk/indications.h, apps/app_readexten.c,
apps/app_disa.c, include/asterisk/channel.h, main/indications.c,
main/channel.c, channels/chan_misdn.c, channels/chan_skinny.c,
funcs/func_channel.c, main/app.c, res/snmp/agent.c,
res/res_indications.c: Merged revisions 168561 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r168561 | russell | 2009-01-13 13:13:05 -0600 (Tue, 13 Jan 2009)
| 2 lines Revert unnecessary indications API change from rev
122314 ........
2009-01-13 17:51 +0000 [r168547] Tilghman Lesher <tlesher@digium.com>
* /, funcs/func_logic.c: Merged revisions 168546 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r168546 | tilghman | 2009-01-13 11:48:00 -0600 (Tue, 13 Jan 2009)
| 6 lines If either conditional is NULL, don't try copying it.
(closes issue #14226) Reported by: caspy Patches:
20090113__bug14226.diff.txt uploaded by Corydon76 (license 14)
........
2009-01-13 16:02 +0000 [r168539] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
* main/taskprocessor.c: correct a CLI description
2009-01-12 23:45 +0000 [r168526] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_alsa.c: Merged revisions 167095 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r167095 | tilghman | 2008-12-31 18:01:22 -0600 (Wed, 31
Dec 2008) | 5 lines Repeat attempts to write when we receive
-EAGAIN from the driver, as detailed in the ALSA sample code (see
http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html#a32)
Reported by: Jerry Geis (via the -users list) Fixed by: me
(license 14) ........
2009-01-12 23:12 +0000 [r168523] Mark Michelson <mmichelson@digium.com>
* main/srv.c: bump the verbosity of a message in srv.c up by one.
It used to be at this level prior to a large patch merge which
converted ast_verbose calls to ast_verb (closes issue #14221)
Reported by: jcovert Patches: srv.c.patch uploaded by jcovert
(license 551)
2009-01-12 23:06 +0000 [r168522] Tilghman Lesher <tlesher@digium.com>
* configure, include/asterisk/autoconfig.h.in, configure.ac,
main/app.c: Some platforms (notably, the BSDs) have a more
efficient implementation called closefrom(3).
2009-01-12 21:51 +0000 [r168508-168517] Jeff Peeler <jpeeler@digium.com>
* /, res/res_agi.c: Merged revisions 168516 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r168516 | jpeeler | 2009-01-12 15:42:34 -0600 (Mon, 12 Jan 2009)
| 5 lines (closes issue #13881) Reported by: hoowa Update the app
CDR field for AGI commands that are not executing an application
via "exec". ........
* /, channels/chan_agent.c: Merged revisions 168507 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r168507 | jpeeler | 2009-01-12 14:26:22 -0600 (Mon, 12
Jan 2009) | 9 lines (closes issue #12269) Reported by: IgorG
Tested by: denisgalvao This gits rid of the notion of an
owning_app allowing the request and hangup to be initiated by
different threads. Originating from an active agent channel
requires this. The implementation primarily changes __login_exec
to wait on a condition variable rather than a lock. Review:
http://reviewboard.digium.com/r/35/ ........
2009-01-12 16:31 +0000 [r168497] Olle Johansson <oej@edvina.net>
* apps/app_minivm.c: Better to use the proper app name
2009-01-12 15:00 +0000 [r168485] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Merged revisions 168482 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r168482 | mmichelson | 2009-01-12 08:58:25 -0600 (Mon, 12 Jan
2009) | 5 lines I am reverting the fix made in revision 168128
(and its upward merges) after being contacted by Olle Johansson
and being shown how this fix is incorrect. Thanks to Olle for
clearing this up for me. ........
2009-01-12 14:57 +0000 [r168481] Russell Bryant <russell@digium.com>
* /, configs/indications.conf.sample: Merged revisions 168480 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r168480 | russell | 2009-01-12 08:57:27 -0600 (Mon, 12 Jan 2009)
| 2 lines s/ringdance/ringcadence/ for Bulgaria ........
2009-01-12 14:35 +0000 [r168479] Olle Johansson <oej@edvina.net>
* main/asterisk.c: Don't include swap.h unless we have swapctl
2009-01-10 01:42 +0000 [r168334] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c: sizeof for a stringfield is 4. Kinda low for
reconstructing a field value.
2009-01-09 23:16 +0000 [r168270] Kevin P. Fleming <kpfleming@digium.com>
* /, sounds/Makefile: Merged revisions 168267 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r168267 | kpfleming | 2009-01-09 17:12:29 -0600 (Fri, 09 Jan
2009) | 1 line update to use new sound file packages that include
license files ........
2009-01-09 23:15 +0000 [r168269] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c: Spacing change
2009-01-09 23:04 +0000 [r168265] Michiel van Baak <michiel@vanbaak.info>
* contrib/scripts/sip_nat_settings (added), CHANGES: Add a script
to find out the correct settings for Asterisk behind NAT (closes
issue #13065) Reported by: tzafrir Patches: sip_nat_settings
uploaded by tzafrir (license 46) sip_nat_settings_6 uploaded by
mvanbaak (license 7) Tested by: tzafrir, pabelanger, Dovid and
moi
2009-01-09 22:21 +0000 [r168200] Russell Bryant <russell@digium.com>
* /, res/res_musiconhold.c: Merged revisions 168198 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r168198 | russell | 2009-01-09 16:14:38 -0600 (Fri, 09
Jan 2009) | 2 lines Make this compile for mvanbaak ........
2009-01-09 21:53 +0000 [r168193] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 168128 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r168128 | mmichelson | 2009-01-09 14:08:04 -0600 (Fri, 09 Jan
2009) | 13 lines Add check_via calls to more request handlers
INFO, NOTIFY, OPTIONS, REFER, and MESSAGE requests were not
checking the topmost Via to determine where to send the response.
Adding check_via calls to those request handlers solves this.
(closes issue #13071) Reported by: baron Patches: check_via.patch
uploaded by baron (license 531) Tested by: baron ........
2009-01-09 21:43 +0000 [r168192] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c, /: Merged revisions 168191 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r168191 | rmudgett | 2009-01-09 15:28:42 -0600 (Fri, 09
Jan 2009) | 3 lines * Fix for JIRA AST-175/ABE-1757 *
Miscellaneous doxygen comments added. ........
2009-01-09 20:25 +0000 [r168142] Terry Wilson <twilson@digium.com>
* res/res_phoneprov.c: Don't leak memory if phoneprov.conf does not
exist (closes issue #14203) Reported by: jamesgolovich Patches:
asterisk-phoneprovleak.diff.txt uploaded by jamesgolovich
(license 176)
2009-01-09 18:30 +0000 [r168090] Tilghman Lesher <tlesher@digium.com>
* res/res_agi.c, include/asterisk/strings.h: When using ast_str
with a non-ast_str-enabled API, we need to update the buffer or
otherwise, we cannot use ast_str_strlen().
2009-01-09 18:01 +0000 [r168014-168054] Matthew Nicholson <mnicholson@digium.com>
* main/logger.c: Added a comment to logger.c about where to put
includes
* main/logger.c: Use ast_safe_system() in logger.c instead of
system() (closes issue #14194) Reported by: pabelanger
2009-01-09 01:15 +0000 [r167935-167973] Terry Wilson <twilson@digium.com>
* apps/app_originate.c: Set ORIGINATE_STATUS instead of
OUTGOING_STATUS to match the documentation
* apps/app_dial.c: Set peer context and exten values so MACRO_EXTEN
and MACRO_CONTEXT will be set
2009-01-08 22:37 +0000 [r167894] Tilghman Lesher <tlesher@digium.com>
* /, res/res_agi.c: Merged revisions 167840 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r167840 | tilghman | 2009-01-08 16:08:56 -0600 (Thu, 08 Jan 2009)
| 6 lines Don't truncate database results at 255 chars. (closes
issue #14069) Reported by: evandro Patches:
20081214__bug14069.diff.txt uploaded by Corydon76 (license 14)
........
2009-01-08 22:34 +0000 [r167888] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Revert chan_sip changes which were
accidentally committed in revision 167792
2009-01-08 21:40 +0000 [r167835-167837] Tilghman Lesher <tlesher@digium.com>
* apps/app_minivm.c: Fix variables to comply with documentation
changes
* apps/app_minivm.c: Textual changes, consistency in status
variable naming, and other minor bugs. (closes issue #13943)
Reported by: Marquis Patches: minivm_trunk_fixes3.patch uploaded
by Marquis (license 32)
2009-01-08 19:48 +0000 [r167792] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c, CHANGES, apps/app_queue.c: Add the average
talk time for a queue This patch adds the functionality to
app_queue of calculating the average amount of time that channels
are bridged for a queue. The algorithm used to calculate the
average is the same exponential average currently used to
calculate the average holdtime. See the CHANGES file to see the
methods you may use to view this information. (closes issue
#13960) Reported by: coolmig Patches:
app_queue.c.diff.trunk-r158840 uploaded by coolmig (license 621)
2009-01-08 19:44 +0000 [r167791] Tilghman Lesher <tlesher@digium.com>
* channels/chan_dahdi.c, CHANGES: Convert dialplan application
DAHDISendCallreroutingFacility to use commas. (closes issue
#13836) Reported by: eliel Patches: chan_dahdi.c.patch uploaded
by eliel (license 64)
2009-01-08 17:26 +0000 [r167700-167720] Kevin P. Fleming <kpfleming@digium.com>
* /, channels/chan_sip.c: Merged revisions 167714 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r167714 | kpfleming | 2009-01-08 11:24:21 -0600 (Thu, 08 Jan
2009) | 1 line remove an unnecessary argument to queue_request()
........
* channels/chan_sip.c: Merged revisions 167620 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r167620 | kpfleming | 2009-01-07 17:32:21 -0600 (Wed, 07 Jan
2009) | 5 lines When a SIP request or response arrives for a
dialog with an associated Asterisk channel, and the lock on that
channel cannot be obtained because it is held by another thread,
instead of dropping the request/response, queue it for later
processing when the channel lock becomes available.
http://reviewboard.digium.com/r/123/ ........
2009-01-08 14:27 +0000 [r167662] Leif Madsen <lmadsen@digium.com>
* contrib/scripts/sip-friends.sql: Oops... fix the fieldname I
changed yesterday to be right.
2009-01-07 22:36 +0000 [r167542-167569] Russell Bryant <russell@digium.com>
* /, main/file.c: Merged revisions 167566 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r167566 | russell | 2009-01-07 16:35:36 -0600 (Wed, 07 Jan 2009)
| 2 lines Fix the last couple of places where free() was
improperly used directly. ........
* /, main/file.c: Merged revisions 167554 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r167554 | russell | 2009-01-07 16:26:42 -0600 (Wed, 07 Jan 2009)
| 2 lines Don't fclose() the file early, the filestream
destructor will handle it. ........
* /, main/file.c: Merged revisions 167545 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r167545 | russell | 2009-01-07 16:19:47 -0600 (Wed, 07 Jan 2009)
| 2 lines Only try to close the file if one was actually opened
........
* /, main/file.c: Merged revisions 167541 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r167541 | russell | 2009-01-07 16:03:59 -0600 (Wed, 07 Jan 2009)
| 4 lines Don't use free() directly. This caused a crash since
ast_filestream is now an ao2 object. Reported by JunK-Y on IRC,
#asterisk-dev ........
2009-01-07 18:20 +0000 [r167478] BJ Weschke <bweschke@btwtech.com>
* apps/app_followme.c: Answer the channel if it has not already
been answered and we've already found a valid profile for
followme. (closes issue #14140) Reported by: dimas Patches:
14140.patch uploaded by dimas
2009-01-07 18:18 +0000 [r167477] Leif Madsen <lmadsen@digium.com>
* configs/queues.conf.sample: Update queues.conf.sample
documentation. Update the queues.conf.sample documentation to
mention that you need to preload chan_local.so as well if you
plan on using Local channels for queue members, and you're
preloading pbx_config.so. (closes issue #14179) Reported by:
CrashHD Tested by: CrashHD
2009-01-07 17:35 +0000 [r167442] Russell Bryant <russell@digium.com>
* /, main/indications.c: Merged revisions 167432 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r167432 | russell | 2009-01-07 11:29:53 -0600 (Wed, 07 Jan 2009)
| 4 lines Treat an empty string the same way as a NULL country
argument. In passing, simplify the handling of returning a
default tone zone. ........
2009-01-07 17:05 +0000 [r167416] Doug Bailey <dbailey@digium.com>
* channels/chan_dahdi.c: Cleanup fsk spill if off hook is detected
during mwi spill. Correct logic error in handling events when
sending mwi spill (closes issue #14143) Reported by: alecdavis
Patches: chan_dahdi.handle_init_event2.diff.txt uploaded by
dbailey
2009-01-07 14:26 +0000 [r167373] Leif Madsen <lmadsen@digium.com>
* contrib/scripts/sip-friends.sql: Update the sip-friends.sql file
to use the non-deprecated 'defaultname' instead of 'username' and
remove an extra comma that would cause the script to fail as-is
2009-01-06 21:36 +0000 [r167301] Mark Michelson <mmichelson@digium.com>
* /, main/db.c: Merged revisions 167299 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r167299 | mmichelson | 2009-01-06 15:35:57 -0600 (Tue, 06 Jan
2009) | 8 lines Use the correct variable when creating the format
string (closes issue #14177) Reported by: nic_bellamy Patches:
asterisk-trunk-svn-r167242-ast_db_gettree.patch uploaded by nic
(license 299) ........
2009-01-06 21:02 +0000 [r167265] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_iax2.c: Merged revisions 167260 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
................ r167260 | tilghman | 2009-01-06 14:48:05 -0600
(Tue, 06 Jan 2009) | 9 lines Merged revisions 167259 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r167259 | tilghman | 2009-01-06 14:44:03 -0600 (Tue, 06
Jan 2009) | 2 lines Security fix AST-2009-001. ........
................
2009-01-05 16:59 +0000 [r167180] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 167179 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r167179 | mmichelson | 2009-01-05 10:51:59 -0600 (Mon, 05 Jan
2009) | 41 lines A couple of changes to T.38 SDP attribute
handling There are some boolean attributes for T.38 such as
T38FaxFillBitRemoval, T38FaxTranscodingMMR, and
T38FaxTranscodingJBIG. By simply being present, we should treat
these as a "true" value. The current code, however, was requiring
a 1 or 0 as the value of the attribute in order to parse it. This
is due to the fact that there are some T.38 endpoints and
gateways that also transmit this information incorrectly. This
patch follows the "be liberal in what you accept and strict in
what you send" philosophy by accepting both the correctly- and
incorrectly-formatted attributes, but only sending information as
it is supposed to be sent. It was also discovered that a
particular type of T.38 gateway sends some non-standard T.38 SDP
attributes. Instead of using T38FaxMaxDatagram and T38MaxBitRate,
it used T38MaxDatagram and T38FaxMaxRate respectively. We now
will properly accept these attributes as well. Note that there
are a lot of patches cited in the below commit message template.
This is because the person who submitted these patches is an
awesome person and wrote 1.4, 1.6.0, and 1.6.1 variants. (closes
issue #13976) Reported by: linulin Patches:
chan_sip.c.1.4-update1.diff uploaded by arcivanov (license 648)
chan_sip.c.1.6.0-update1.diff uploaded by arcivanov (license 648)
chan_sip.c.1.6.1-update1.diff uploaded by arcivanov (license 648)
chan_sip.c.1.4-relaxedT38_update1.diff uploaded by arcivanov
(license 648) chan_sip.c.1.6.0-relaxedT38_update1.diff uploaded
by arcivanov (license 648)
chan_sip.c.1.6.1-relaxedT38_update1.diff uploaded by arcivanov
(license 648) Tested by: arcivanov ........
2009-01-05 16:44 +0000 [r167176] Tilghman Lesher <tlesher@digium.com>
* UPGRADE-1.6.txt: More clearly explain that quote marks are no
longer necessary. (closes issue #13718) Reported by: davidw
Patches: 20081020__bug13718.diff.txt uploaded by Corydon76
(license 14) Tested by: blitzrage
2009-01-03 20:29 +0000 [r167125] Jeff Peeler <jpeeler@digium.com>
* main/asterisk.c: When parsing environment variable
ASTERISK_PROMPT, make sure to proceed to the next character when
a non format specifier is used (no %). Otherwise, the while loop
looking for the null byte will never exit.
2008-12-31 23:07 +0000 [r167061] Sean Bright <sean.bright@gmail.com>
* doc/CODING-GUIDELINES, include/asterisk.h, channels/h323/README:
Mostly just whitespace, but also convert 'CVS' to 'SVN' in a
couple places and fix a few typos I found in the
CODING_GUIDELINES.
2008-12-31 22:53 +0000 [r167057] Terry Wilson <twilson@digium.com>
* main/xmldoc.c: Don't forget to free typename
2008-12-31 21:52 +0000 [r167021] Mark Michelson <mmichelson@digium.com>
* channels/chan_dahdi.c: Change some incorrect syntax for pri set
debug and correct an off-by-one error in ss7 set debug command
2008-12-31 19:39 +0000 [r166954-166958] Tilghman Lesher <tlesher@digium.com>
* main/ast_expr2.h, main/ast_expr2.c: That was weird...
* channels/chan_local.c, /, main/ast_expr2.h, main/ast_expr2.c:
Merged revisions 166953 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r166953 | tilghman | 2008-12-31 13:20:35 -0600 (Wed, 31 Dec 2008)
| 5 lines Also inherit the musiconhold class. (Closes #14153)
Reported by: Jerry Geis, via the users list. Patch by: me
(license 14) ........
2008-12-30 20:50 +0000 [r166908] Terry Wilson <twilson@digium.com>
* res/res_phoneprov.c, doc/sip-retransmit.txt,
doc/tex/phoneprov.tex, res/res_http_post.c,
phoneprov/polycom_line.xml, doc/realtimetext.txt: Fix some
svn:keywords
2008-12-29 18:04 +0000 [r166861] Mark Michelson <mmichelson@digium.com>
* apps/app_dial.c, apps/app_queue.c: Update app_queue to deal with
the removal of AST_PBX_KEEPALIVE When placing a call to a queue
which ran a gosub on the member's channel, Asterisk would crash
every time, stemming from the fact that the member's channel was
being hung up unexpectedly when the Gosub completed. The
necessary change was pretty much copied and pasted from
app_dial's similar changes made last week. I also took the
opportunity to change a LOG_DEBUG message in app_dial to use
ast_debug. I am guessing this was due to a direct merge from 1.4
that was not corrected to use trunk's preferred syntax.
2008-12-28 15:36 +0000 [r166823] Eliel C. Sardanons <eliels@gmail.com>
* funcs/func_audiohookinherit.c: Fix a typo in the XML
documentation of the AUDIOHOOK_INHERIT dialplan function.
2008-12-28 15:15 +0000 [r166773] Russell Bryant <russell@digium.com>
* /, channels/misdn_config.c: Merged revisions 166772 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r166772 | russell | 2008-12-28 09:13:48 -0600 (Sun, 28
Dec 2008) | 4 lines Use strncat() instead of an sprintf() in
which source and target buffers overlap
http://lists.digium.com/pipermail/asterisk-dev/2008-December/035919.html
........
2008-12-24 15:10 +0000 [r166731] Terry Wilson <twilson@digium.com>
* channels/chan_sip.c: There is no section 22.2.2 in rfc 3261. I
believe 26.2.2 is what was meant: Note that in the SIPS URI
scheme, transport is independent of TLS, and thus
"sips:alice@atlanta.com;transport=tcp" and
"sips:alice@atlanta.com;transport=sctp" are both valid (although
note that UDP is not a valid transport for SIPS). The use of
"transport=tls" has consequently been deprecated, partly because
it was specific to a single hop of the request. This is a change
since RFC 2543.
2008-12-23 20:47 +0000 [r166696] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c: Allow semicolons and extended characters in
user-specified SIP headers. (closes issue #14110) Reported by:
gork Patches: 20081222__bug14110__2.diff.txt uploaded by
Corydon76 (license 14) Tested by: gork, putnopvut
2008-12-23 18:13 +0000 [r166665] Steve Murphy <murf@digium.com>
* apps/app_dial.c, main/pbx.c, /, main/features.c,
apps/app_macro.c, include/asterisk/pbx.h, apps/app_queue.c,
include/asterisk/features.h: Merged revisions 166093 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4 In
order to merge this 1.4 patch into trunk, I had to resolve some
conflicts and wait for Russell to make some changes to res_agi. I
re-ran all the tests; 39 calls in all, and made fairly careful
notes and comparisons: I don't want this to blow up some aspect
of asterisk; I completely removed the KEEPALIVE from the pbx.h
decls. The first 3 scenarios involving feature park; feature xfer
to 700; hookflash park to Park() app call all behave the same,
don't appear to leave hung channels, and no crashes. ........
r166093 | murf | 2008-12-19 15:30:32 -0700 (Fri, 19 Dec 2008) |
131 lines This merges the masqpark branch into 1.4 These changes
eliminate the need for (and use of) the KEEPALIVE return code in
res_features.c; There are other places that use this result code
for similar purposes at a higher level, these appear to be left
alone in 1.4, but attacked in trunk. The reason these changes are
being made in 1.4, is that parking ends a channel's life, in some
situations, and the code in the bridge (and some other places),
was not checking the result code properly, and dereferencing the
channel pointer, which could lead to memory corruption and
crashes. Calling the masq_park function eliminates this danger in
higher levels. A series of previous commits have replaced some
parking calls with masq_park, but this patch puts them ALL to
rest, (except one, purposely left alone because a masquerade is
done anyway), and gets rid of the code that tests the KEEPALIVE
result, and the NOHANGUP_PEER result codes. While bug 13820
inspired this work, this patch does not solve all the problems
mentioned there. I have tested this patch (again) to make sure I
have not introduced regressions. Crashes that occurred when a
parked party hung up while the parking party was listening to the
numbers of the parking stall being assigned, is eliminated. These
are the cases where parking code may be activated: 1. Feature one
touch (eg. *3) 2. Feature blind xfer to parking lot (eg ##700) 3.
Run Park() app from dialplan (eg sip xfer to 700) (eg. dahdi
hookflash xfer to 700) 4. Run Park via manager. The interesting
testing cases for parking are: I. A calls B, A parks B a. B hangs
up while A is getting the numbers announced. b. B hangs up after
A gets the announcement, but before the parking time expires c. B
waits, time expires, A is redialed, A answers, B and A are
connected, after which, B hangs up. d. C picks up B while still
in parking lot. II. A calls B, B parks A a. A hangs up while B is
getting the numbers announced. b. A hangs up after B gets the
announcement, but before the parking time expires c. A waits,
time expires, B is redialed, B answers, A and B are connected,
after which, A hangs up. d. C picks up A while still in parking
lot. Testing this throroughly involves acting all the
permutations of I and II, in situations 1,2,3, and 4. Since I
added a few more changes (ALL references to KEEPALIVE in the
bridge code eliimated (I missed one earlier), I retested most of
the above cases, and no crashes. H-extension weirdness. Current
h-extension execution is not completely correct for several of
the cases. For the case where A calls B, and A parks B, the 'h'
exten is run on A's channel as soon as the park is accomplished.
This is expected behavior. But when A calls B, and B parks A,
this will be current behavior: After B parks A, B is hung up by
the system, and the 'h' (hangup) exten gets run, but the channel
mentioned will be a derivative of A's... Thus, if A is DAHDI/1,
and B is DAHDI/2, the h-extension will be run on channel
Parked/DAHDI/1-1<ZOMBIE>, and the start/answer/end info will be
those relating to Channel A. And, in the case where A is
reconnected to B after the park time expires, when both parties
hang up after the joyful reunion, no h-exten will be run at all.
In the case where C picks up A from the parking lot, when either
A or C hang up, the h-exten will be run for the C channel. CDR's
are a separate issue, and not addressed here. As to WHY this
strange behavior occurs, the answer lies in the procedure
followed to accomplish handing over the channel to the parking
manager thread. This procedure is called masquerading. In the
process, a duplicate copy of the channel is created, and most of
the active data is given to the new copy. The original channel
gets its name changed to XXX<ZOMBIE> and keeps the PBX
information for the sake of the original thread (preserving its
role as a call originator, if it had this role to begin with),
while the new channel is without this info and becomes a call
target (a "peer"). In this case, the parking lot manager thread
is handed the new (masqueraded) channel. It will not run an
h-exten on the channel if it hangs up while in the parking lot.
The h exten will be run on the original channel instead, in the
original thread, after the bridge completes. See bug 13820 for
our intentions as to how to clean up the h exten behavior.
Review: http://reviewboard.digium.com/r/29/ ........
2008-12-23 16:04 +0000 [r166625] Russell Bryant <russell@digium.com>
* CHANGES: Fix spelling error.
2008-12-23 15:17 +0000 [r166569] Mark Michelson <mmichelson@digium.com>
* main/channel.c, /: Merged revisions 166568 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r166568 | mmichelson | 2008-12-23 09:16:26 -0600 (Tue, 23 Dec
2008) | 12 lines Fix a crash resulting from a datastore with
inheritance but no duplicate callback The fix for this is to
simply set the newly created datastore's data pointer to NULL if
it is inherited but has no duplicate callback. (closes issue
#14113) Reported by: francesco_r Patches: 14113.patch uploaded by
putnopvut (license 60) Tested by: francesco_r ........
2008-12-23 04:32 +0000 [r166533] Tilghman Lesher <tlesher@digium.com>
* main/channel.c, /: Merged revisions 166509 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r166509 | tilghman | 2008-12-22 22:05:25 -0600 (Mon, 22 Dec 2008)
| 4 lines Use the integer form of condition for integer
comparisons. (closes issue #14127) Reported by: andrew ........
2008-12-22 23:25 +0000 [r166470] Mark Michelson <mmichelson@digium.com>
* res/res_agi.c: Always use the value of the AGISIGHUP when running
an AGI. Prior to this patch, the value of AGISIGUP was not always
honored when set on a channel. (closes issue #13711) Reported by:
fmueller Patches: 13711.patch uploaded by putnopvut (license 60)
2008-12-22 21:45 +0000 [r166436] Russell Bryant <russell@digium.com>
* res/res_musiconhold.c: Cosmetic change - don't mix struct
initializer styles.
2008-12-22 21:08 +0000 [r166382] Mark Michelson <mmichelson@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 166380 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r166380 | mmichelson | 2008-12-22 14:56:29 -0600 (Mon,
22 Dec 2008) | 36 lines Fix a deadlock relating to channel locks
and autoservice It has been discovered that if a channel is
locked prior to a call to ast_autoservice_stop, then it is likely
that a deadlock will occur. The reason is that the call to
ast_autoservice_stop has a check built into it to be sure that
the thread running autoservice is not currently trying to
manipulate the channel we are about to pull out of autoservice.
The autoservice thread, however, cannot advance beyond where it
currently is, though, because it is trying to acquire the lock of
the channel for which autoservice is attempting to be stopped.
The gist of all this is that a channel MUST NOT be locked when
attempting to stop autoservice on the channel. In this particular
case, the channel was locked by a call to ast_read. A call to
ast_exists_extension led to autoservice being started and stopped
due to the existence of dialplan switches. It may be that there
are future commits which handle the same symptoms but in a
different location, but based on my looks through the code, it is
very rare to see a construct such as this one. (closes issue
#14057) Reported by: rtrauntvein Patches: 14057v3.patch uploaded
by putnopvut (license 60) Tested by: rtrauntvein Review:
http://reviewboard.digium.com/r/107/ ........
2008-12-22 20:26 +0000 [r166273-166377] Russell Bryant <russell@digium.com>
* res/res_musiconhold.c: Fix a bad typo.
* main/astobj2.c: Remove some error messages. This is the default
handler that is valid to use.
* /, main/utils.c: Merged revisions 166297 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r166297 | russell | 2008-12-22 11:22:56 -0600 (Mon, 22 Dec 2008)
| 2 lines Fix up timeout handling in ast_carefulwrite(). ........
* include/asterisk/utils.h, main/manager.c, main/utils.c: Introduce
ast_careful_fwrite() and use in AMI to prevent partial writes.
This patch introduces a function to do careful writes on a file
stream which will handle timeouts and partial writes. It is
currently used in AMI to address the issue that has been
reported. However, there are probably a few other places where
this could be used. (closes issue #13546) Reported by: srt Tested
by: russell http://reviewboard.digium.com/r/104/
* res/res_musiconhold.c: Re-work ref count handling of MoH classes
using astobj2 to resolve crashes. (closes issue #13566) Reported
by: igorcarneiro Tested by: russell Review:
http://reviewboard.digium.com/r/106/
2008-12-22 16:08 +0000 [r166268] Joshua Colp <jcolp@digium.com>
* main/dnsmgr.c: Record the previous port in the temporary address
structure so that the comparison does not treat the host as
having changed even if it did not. This would have been
uninitialized before and would have led to a baddddd port.
(closes issue #13628) Reported by: pananix Patches:
bug13628.patch uploaded by jpeeler (license 325) Tested by: file,
blitzrage
2008-12-22 16:07 +0000 [r166267] Mark Michelson <mmichelson@digium.com>
* funcs/func_timeout.c, main/file.c: Fix a file playback crash and
explicitly initialize values in func_timeout.c A crash was
brought up on the bugtracker. The first run through valgrind was
full of legitimate complaints of uninitialized values in
func_timeout when setting a response timeout. These were fixed
but the crash persisted. A second run through showed the real
problem. The reference counting used for filestreams was
incorrect because there were some missing increments when a frame
was read from a format module. (closes issue #14118) Reported by:
blitzrage Patches: 14118v2.patch uploaded by putnopvut (license
60) Tested by: blitzrage
2008-12-22 14:16 +0000 [r166258] Russell Bryant <russell@digium.com>
* res/res_agi.c: Remove AST_PBX_KEEPALIVE usage from res_agi. This
patch removes the usage of AST_PBX_KEEPALIVE from res_agi. The
only usage was for the AGI command, "asyncagi break". This patch
removes this feature. Normally, a feature would not be removed
like this. However, this code is broken and usage of it will
result in a memory leak. Usage of this feature will make the AGI
code return a result of AST_PBX_KEEPALIVE. The PBX handler
assumes that another thread has assumed ownership of the channel.
The channel thread will exit without destroying the channel.
Unfortunately, _no_ thread has ownership of the channel at this
point. There are a couple of serious problems here: 1) The only
way to recover the caller is to issue a channel redirect. This
will work, but this will be done with a masquerade, and the old
ast_channel structure will be lost. 2) Until the channel redirect
happens, there is no code servicing the channel. That means
nothing is reading audio or handling events coming from the
channel. This is very bad. The recommended way to get this same
"break" functionality is to issue the redirect while the channel
is still being handled by the AGI code. That way, there will be
no memory leak, and there will be no period of time that the
channel is not being serviced.
2008-12-20 01:37 +0000 [r166219] Russell Bryant <russell@digium.com>
* include/asterisk/doxyref.h: Make a note about formatting the
review URL in commit messages
2008-12-19 23:45 +0000 [r166092-166162] Mark Michelson <mmichelson@digium.com>
* main/audiohook.c: Get rid of an extra space. I don't know how
this crept back in when I had already fixed it earlier
* funcs/func_audiohookinherit.c: Remove the verbatim tag from the
author line I could have sworn I already did that before,
though...
* main/channel.c, funcs/func_audiohookinherit.c (added),
include/asterisk/audiohook.h, main/audiohook.c, CHANGES: Adding a
new dialplan function AUDIOHOOK_INHERIT This function is being
added as a method to allow for an audiohook to move to a new
channel during a channel masquerade. The most obvious use for
such a facility is for MixMonitor when a transfer is performed.
Prior to the addition of this functionality, if a channel running
MixMonitor was transferred by another party, then the recording
would stop once the transfer had completed. By using
AUDIOHOOK_INHERIT, you can make MixMonitor continue recording the
call even after the transfer has completed. It has also been
determined that since this is seen by most as a bug fix and is
not an invasive change, this functionality will also be
backported to 1.4 and merged into the 1.6.0 branches, even though
they are feature-frozen. (closes issue #13538) Reported by: mbit
Patches: 13538.patch uploaded by putnopvut (license 60) Tested
by: putnopvut Review: http://reviewboard.digium.com/r/102/
2008-12-19 21:44 +0000 [r166058] Matthew Fredrickson <creslin@digium.com>
* channels/chan_dahdi.c, configure,
include/asterisk/autoconfig.h.in, configure.ac: Add configuration
support for half_full DAHDI buffer policy
2008-12-19 18:20 +0000 [r165954] Eliel C. Sardanons <eliels@gmail.com>
* apps/app_record.c: Fix the XML documentation for Record().
<value> tags inside <variable> elements must have CDATA and no
another XML node.
2008-12-19 15:05 +0000 [r165801-165890] Russell Bryant <russell@digium.com>
* /, apps/app_chanspy.c: Merged revisions 165889 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r165889 | russell | 2008-12-19 09:03:02 -0600 (Fri, 19 Dec 2008)
| 9 lines Ensure that the chanspy datastore is fully initialized.
This patch resolved some random crash issues observed by a user
on a BSD system (closes issue #14111) Reported by: ys Patches:
app_chanspy.c.diff uploaded by ys (license 281) ........
* include/asterisk/doxyref.h: Disable some automatic links
generated by doxygen.
* include/asterisk/doxyref.h: Introduce commit message formatting
guidelines. This documents the recommended outline to use for
commit message. It also covers information on special tags that
can be used in commit messages, as well as the template
functionality that is available on bugs.digium.com. Review:
http://reviewboard.digium.com/r/96/
* /, main/utils.c: Merged revisions 165796 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r165796 | russell | 2008-12-18 15:39:25 -0600 (Thu, 18 Dec 2008)
| 11 lines Make ast_carefulwrite() be more careful. This patch
handles some additional cases that could result in partial writes
to the file description. This was done to address complaints
about partial writes on AMI. (issue #13546) (more changes needed
to address potential problems in 1.6) Reported by: srt Tested by:
russell Review: http://reviewboard.digium.com/r/99/ ........
2008-12-18 21:43 +0000 [r165798] Jeff Peeler <jpeeler@digium.com>
* main/manager.c: (closes issue #13993) Reported by: mika Add
ActionID response to ping if sent with request.
2008-12-18 21:41 +0000 [r165797] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_voicemail.c: Merged revisions 165767 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r165767 | tilghman | 2008-12-18 15:14:47 -0600 (Thu, 18
Dec 2008) | 8 lines Add mutexes around accesses to the IMAP
library interface. This prevents certain crashes, especially when
shared mailboxes are used. (closes issue #13653) Reported by:
howardwilkinson Patches:
asterisk-1.4.21.2-appvoicemail-sharedimap-lock.patch uploaded by
howardwilkinson (license 590) Tested by: jpeeler ........
2008-12-18 21:21 +0000 [r165792] Joshua Colp <jcolp@digium.com>
* channels/chan_dahdi.c, channels/chan_misdn.c,
channels/chan_sip.c, pbx/pbx_ael.c, apps/app_queue.c,
channels/chan_oss.c: Numerous documentation updates. (closes
issue #13970) Reported by: pkempgen Patches:
__20081217_cli_usage_fixes.patch.txt uploaded by blitzrage
(license 10)
2008-12-18 19:34 +0000 [r165724] Mark Michelson <mmichelson@digium.com>
* res/res_odbc.c: Fix crashes in res_odbc. The variable "class" was
being set NULL just prior to being dereferenced in an ao2_link
call. I have moved the setting of the variable to NULL until
after the ao2_link call.
2008-12-18 19:33 +0000 [r165662-165723] Russell Bryant <russell@digium.com>
* apps/app_dial.c, main/pbx.c, include/asterisk/pbx.h: Remove the
need for AST_PBX_KEEPALIVE with the GoSub option from Dial. This
is part of an effort to completely remove AST_PBX_KEEPALIVE and
other similar return codes from the source. While this usage was
perfectly safe, there are others that are problematic. Since we
know ahead of time that we do not want to PBX to destroy the
channel, the PBX API has been changed so that information can be
provided as an argument, instead, thus removing the need for the
KEEPALIVE return value. Further changes to get rid of KEEPALIVE
and related code is being done by murf. There is a patch up for
that on review 29. Review: http://reviewboard.digium.com/r/98/
* /, res/res_musiconhold.c: Merged revisions 165661 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r165661 | russell | 2008-12-18 12:52:18 -0600 (Thu, 18
Dec 2008) | 7 lines Set the process group ID on the MOH process
so that all children will get killed (closes issue #14099)
Reported by: caspy Patches: res_musiconhold.c.patch.killpg.try2
uploaded by caspy (license 645) ........
2008-12-18 18:36 +0000 [r165658] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: Fix 2 resource leaks and fix another
pipe-to-comma conversion
2008-12-18 17:13 +0000 [r165599] Joshua Colp <jcolp@digium.com>
* /, main/rtp.c: Merged revisions 165591 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r165591 | file | 2008-12-18 13:11:42 -0400 (Thu, 18 Dec 2008) | 4
lines Only care about a compatible codec for early bridging if we
are actually bridging to another channel. If we are not we
actually want to bring the audio back to us. (closes issue
#13545) Reported by: davidw ........
2008-12-18 16:36 +0000 [r165541] Tilghman Lesher <tlesher@digium.com>
* res/res_odbc.c: Fix reference counts of the class and add an
assertion to the end.
2008-12-18 15:25 +0000 [r165502] Eliel C. Sardanons <eliels@gmail.com>
* main/strings.c, include/asterisk/strings.h: Remove duplicate code
from the ast_str API. We now use __AST_STR_* to access 'struct
ast_str' members, but this must only be used inside the API
implementation. (closes issue #14098) Reported by: eliel Patches:
ast_str.patch uploaded by eliel (license 64)
2008-12-18 14:23 +0000 [r165433-165469] Russell Bryant <russell@digium.com>
* apps/app_originate.c: Add a \todo note for app_originate. Jared
Smith suggested that we add a way to be able to set variables and
functions on the outbound channel. I think that it's a great
idea, so I have added it as a todo so that it gets done at some
point.
* apps/app_originate.c (added), CHANGES: Add a new application,
Originate. (closes issue #14075) Reported by: rcasas Patches:
app_originate.c uploaded by rcasas (license 641), heavily
modified by me Tested by: russell Review:
http://reviewboard.digium.com/r/95/
2008-12-17 23:39 +0000 [r165397] Tilghman Lesher <tlesher@digium.com>
* apps/app_record.c: Add RECORD_STATUS variable, as requested on
the -users list. Patch by me (license 14)
2008-12-17 21:46 +0000 [r165326-165330] Mark Michelson <mmichelson@digium.com>
* res/res_odbc.c: Fix a refcount leak in res_odbc
* apps/app_meetme.c, res/res_realtime.c: Fix the build
2008-12-17 21:28 +0000 [r165319-165325] Tilghman Lesher <tlesher@digium.com>
* apps/app_macro.c: Oops, broke trunk
* /, apps/app_macro.c: Merged revisions 165317 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r165317 | tilghman | 2008-12-17 15:14:37 -0600 (Wed, 17 Dec 2008)
| 4 lines Reverse the fix from issue #6176 and add proper
handling for that issue. (Closes issue #13962, closes issue
#13363) Fixed by myself (license 14) ........
2008-12-17 21:17 +0000 [r165318] Mark Michelson <mmichelson@digium.com>
* apps/app_meetme.c, res/res_realtime.c, apps/app_directory.c,
apps/app_queue.c, apps/app_voicemail.c: Merged revisions 165255
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r165255 | mmichelson | 2008-12-17 14:51:38 -0600 (Wed, 17 Dec
2008) | 7 lines Fix some memory leaks found while looking at how
realtime configs are handled. Also cleaned up some coding
guidelines violations in app_realtime.c, mostly related to
spacing ........
2008-12-17 20:50 +0000 [r165254] Steve Murphy <murf@digium.com>
* utils/extconf.c: This patch is here committed to satisfy the
buildbot, who has a problem with the const.
2008-12-17 19:55 +0000 [r165219] Terry Wilson <twilson@digium.com>
* res/res_phoneprov.c: Polycom phones close the connection after
reading a little bit of the firmware files, we should stop
sending in that case. Also, make that case print out a debug
statement instead of a scary WARNING.
2008-12-17 19:52 +0000 [r165216] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Call proxy_update so that the IP address
gets populated. Sending stuff to 0.0.0.0 is silly! (closes issue
#14055) Reported by: chris-mac
2008-12-17 18:49 +0000 [r165180] Matthew Nicholson <mnicholson@digium.com>
* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: This patch
adds a new 'ignoresdpversion' option to sip.conf. When this is
enabled (either globally or for a specific peer), chan_sip will
treat any SDP data it receives as new data and update the media
stream accordingly. By default, Asterisk will only modify the
media stream if the SDP session version received is different
from the current SDP session version. This option is required to
interoperate with devices that have non-standard SDP session
version implementations (observed by toc on the bug tracker with
Microsoft OCS which always uses 0 as the session version).
http://reviewboard.digium.com/r/94/ (closes issue #13958)
Reported by: toc Tested by: toc
2008-12-17 17:56 +0000 [r165145] Russell Bryant <russell@digium.com>
* doc/appdocsxml.dtd: argsep is used as an attribute for an
argument, as well
2008-12-17 17:53 +0000 [r165142-165143] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: And actually assign the function to a
pointer...
* apps/app_voicemail.c: Use the create_vm_state_from_user function
in a place where it was not being used before. Also, I've moved
the urgent folder check in messagecount() up a bit so that the
flow is a bit better. This was something I noticed while taking a
look at issue #13973, although I don't think this is the
underlying cause of the issue.
2008-12-17 16:41 +0000 [r165108] Kevin P. Fleming <kpfleming@digium.com>
* utils: ignore this copied file
2008-12-17 05:04 +0000 [r165039-165071] Steve Murphy <murf@digium.com>
* utils/Makefile, pbx/pbx_ael.c, utils/ael_main.c, utils/extconf.c,
utils/conf2ael.c, utils/check_expr.c: A possibly "horrible fix"
for a "horribly broken" situation. As stuff shifts around in the
asterisk code, the miscellaneous inclusions from the standalone
stuff gets broken. There's no easy fix for this situation. I made
sure that everything in utils builds without problem ***AND***
that aelparse runs the regressions correctly with the following
make menuselect options both on and off: DONT_OPTIMIZE
DEBUG_THREADS DEBUG_CHANNEL_LOCKS MALLOC_DEBUG MTX_PROFILE
DEBUG_SCHEDULER DEBUG_THREADLOCALS DETECT_DEADLOCKS CHANNEL_TRACE
I think from now on, I'm going to #undef all these features in
the various utils native files; I guess I could do the same for
the copied-in files, surrounded by STANDALONE ifdef. A standalone
isn't going to care about threads, mutexes, etc.
* pbx/ael/ael-test/ref.ael-vtest17,
pbx/ael/ael-test/ref.ael-vtest13: fixed the regressions
2008-12-16 23:06 +0000 [r164978] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 164977 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r164977 | mmichelson | 2008-12-16 17:04:27 -0600 (Tue, 16 Dec
2008) | 7 lines After looking through SIP registration code most
of the day, this is one of the few things I could find that was
just plain wrong. Even though it probably isn't possible for it
to happen, it seems weird to have code that checks if a pointer
is NULL and then immediately dereferences that pointer if it was
NULL. ........
2008-12-16 22:57 +0000 [r164976] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c, doc/api-1.6.2-changes.txt (added),
funcs/func_logic.c, include/asterisk/pbx.h, utils/extconf.c,
CHANGES, configs/extensions.conf.sample: Add timezone to the
possible fields in a timespec. (closes issue #14028) Reported by:
mostyn Patches: timezone-v2.patch uploaded by mostyn (license
398) (with additional code guideline fixes and a memory leak fix
by me - license 14)
2008-12-16 22:45 +0000 [r164942] Jeff Peeler <jpeeler@digium.com>
* apps/app_record.c: (closes issue #13669) Reported by: pj Delete
file recording if recording terminated from a hangup.
2008-12-16 22:31 +0000 [r164941] Terry Wilson <twilson@digium.com>
* channels/chan_sip.c: Make a note of the feature request in bug
#11157 as per the reporter and oej, and suspend the bug since no
one seems to be keen on implementing it any time soon.
2008-12-16 21:39 +0000 [r164821-164882] Russell Bryant <russell@digium.com>
* /, main/utils.c: Merged revisions 164881 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r164881 | russell | 2008-12-16 15:38:29 -0600 (Tue, 16 Dec 2008)
| 9 lines Fix an issue where DEBUG_THREADS may erroneously report
that a thread is exiting while holding a lock. If the last lock
attempt was a trylock, and it failed, it will still be in the
list of locks so that it can be reported. (closes issue #13219)
Reported by: pj ........
* /, apps/app_macro.c: Merged revisions 164876 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r164876 | russell | 2008-12-16 15:10:44 -0600 (Tue, 16 Dec 2008)
| 6 lines Do not dereference the channel if AST_PBX_KEEPALIVE has
been returned. This is a bug I noticed while looking at the code
for app_macro. This return code means that another thread has
assumed ownership of the channel and it can no longer be touched.
(I hate this return code with a passion, by the way.) ........
* main/asterisk.c: Fix build issues on Linux after sysinfo related
changes
2008-12-16 20:47 +0000 [r164809-164814] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Qualify
trumps poke per lmadsen.
* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add
configuration options for finer control over how Asterisk handles
having to poke all peers at seemingly the same time. (closes
issue #13217) Reported by: cervajs
2008-12-16 20:41 +0000 [r164807] Russell Bryant <russell@digium.com>
* main/manager.c, /: Merged revisions 164806 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r164806 | russell | 2008-12-16 14:35:25 -0600 (Tue, 16 Dec 2008)
| 9 lines Add "restart gracefully" to the AMI blacklist of CLI
commands. "module unload" was already identified as a command
that can not be used from the AMI. "restart gracefully"
effectively unloads all modules, and will run in to the same
problems. (closes issue #13894) Reported by: kernelsensei
........
2008-12-16 20:08 +0000 [r164802] Michiel van Baak <michiel@vanbaak.info>
* configure, include/asterisk/autoconfig.h.in, configure.ac,
main/asterisk.c: introduce 'core show sysinfo' for systems that
dont have the Linux-ish sysinfo stuff but do have sysctl. (closes
issue #13433) Reported by: mvanbaak Patches:
2008121300_sysinfosysctl.diff.txt uploaded by mvanbaak (license
7) with two free calls replaced with ast_free based on feedback
on reviewboard Review: http://reviewboard.digium.com/r/91/
2008-12-16 20:04 +0000 [r164801] Steve Murphy <murf@digium.com>
* main/pbx.c: (closes issue #14076) Reported by: toc Tested by:
murf OK, Well this issue has had its share of flip-flopping. I
found the following: 1. the code in question, in ext_cmp1 in
pbx.c, would not allow two extensions that vary only by any
dashes contained within them, to be defined in the same context.
2. for input dialstrings, dashes are NOT ignored. So, skipping
them when sorting patterns seemed a bit silly. Thus, you might
declare ext 891 in a context, but if you try dialing 8-9-1, it
will NOT match 891. So, I proposed to remove the code from
ext_cmp1 to skip the spaces and dashes. Just kept us from
declaring 891 and 8-9-1 in the same context, forcing users to
generate otherwise uselessly obfuscated dialplan code to get the
same effect. Then, I tried out 1.4, and found that: 1. you can
declare 891 and 8-9-1 in the same context! 2. You can't define
891, and have 8-9-1 match it! Nor can you define 8-9-1, and have
891 match it! So, it appears that my proposal simply restores the
pbx to behaving as it did in 1.4.
2008-12-16 19:54 +0000 [r164798] Tilghman Lesher <tlesher@digium.com>
* contrib/scripts/safe_asterisk: Set up umask as a possible
configuration option. (closes issue #13753) Reported by: irroot
2008-12-16 17:14 +0000 [r164737] Russell Bryant <russell@digium.com>
* /, main/threadstorage.c, include/asterisk/threadstorage.h: Merged
revisions 164736 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r164736 | russell | 2008-12-16 11:06:29 -0600 (Tue, 16 Dec 2008)
| 14 lines Fix memory leak and invalid reporting issues with
DEBUG_THREADLOCALS. One issue was that the ast_mutex_* API was
being used within the context of the thread local data
destructors. We would go off and allocate more thread local data
while the pthread lib was in the middle of destroying it all.
This led to a memory leak. Another issue was an invalid argument
being provided to the the object_add API call. (closes issue
#13678) Reported by: ys Tested by: Russell ........
2008-12-16 16:50 +0000 [r164733] Joshua Colp <jcolp@digium.com>
* pbx/pbx_config.c: Be more detailed about why the include did not
get included. (closes issue #14071) Reported by: kshumard
Patches: pbx_config.patch.improvederroroutput.txt uploaded by
kshumard (license 92)
2008-12-16 16:00 +0000 [r164675] Russell Bryant <russell@digium.com>
* /, channels/chan_sip.c: Merged revisions 164672 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r164672 | russell | 2008-12-16 09:56:37 -0600 (Tue, 16 Dec 2008)
| 11 lines Fix a memory leak related to the use of the "setvar"
configuration option. The problem was that these variables were
being appended to the list of vars on the sip_pvt every time a
re-registration or re-subscription came in. Since it's just a
waste of memory to put them there unless the request was an
INVITE, then the fix is to check the request type before copying
the vars. (closes issue #14037) Reported by: marvinek Tested by:
russell ........
2008-12-16 15:44 +0000 [r164659] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: When using externhost make sure the port
gets set to the bindaddr port if one was not specified in the
externhost value itself. (closes issue #13634) Reported by:
performer
2008-12-16 15:31 +0000 [r164648] Steve Murphy <murf@digium.com>
* main/pbx.c, /: Merged revisions 164634 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r164634 | murf | 2008-12-16 08:15:58 -0700 (Tue, 16 Dec 2008) | 5
lines I added a sentence to clarify why - and ' ' are ignored in
patterns as per bug 14076. Leif says he'll put some stuff about
it in the extensions.conf sample, etc. ........
2008-12-16 15:00 +0000 [r164602-164623] Russell Bryant <russell@digium.com>
* apps/app_minivm.c: Set MINIVM_ACCMESS_STATUS in all cases. Also,
remove a variable that was not needed. (closes issue #14081)
Reported by: pkempgen
* /, res/res_musiconhold.c: Merged revisions 164605 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r164605 | russell | 2008-12-16 08:28:10 -0600 (Tue, 16
Dec 2008) | 5 lines Don't try to change working directory if a
directory was not configured. (closes issue #14089) Reported by:
caspy ........
* channels/chan_dahdi.c: Fix usage of the DAHDI_VMWI ioctl. (closes
issue #14090) Reported by: alecdavis Patches:
chan_dahdi.VMWI_ioctl.diff.txt uploaded by alecdavis (license
585)
2008-12-16 01:52 +0000 [r164565] Sean Bright <sean.bright@gmail.com>
* doc/tex/odbcstorage.tex: Use tables instead of ASCII art. Also
change a bit of minor formatting.
2008-12-15 22:25 +0000 [r164519-164525] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Open a timer before loading configuration
so that the trunking configuration option will take effect.
(closes issue #14082) Reported by: seandarcy
* channels/chan_iax2.c: Fix log message to refer to the generic
timing interface, not DAHDI specifically (inspired by issue
#14082)
* main/frame.c: Make sure we handle a uint32_t payload in
ast_frdup() (closes issue #14080) Reported by: fnordian Patches:
frame.patch uploaded by fnordian (license 110)
2008-12-15 21:17 +0000 [r164485] Tilghman Lesher <tlesher@digium.com>
* configs/extconfig.conf.sample, pbx/pbx_realtime.c, CHANGES: Allow
disabling pattern match searches within the Realtime dialplan
switch. (closes issue #13698) Reported by: fhackenberger Patches:
20081211__bug13698.diff.txt uploaded by Corydon76 (license 14)
Tested by: fhackenberger
2008-12-15 20:07 +0000 [r164419-164428] Mark Michelson <mmichelson@digium.com>
* apps/app_page.c: Add an 'i' option to app_page. This option works
the same as the 'i' options for app_dial and app_queue, in that
they will ignore any attempts by phones to forward the call.
(closes issue #13977) Reported by: putnopvut Patches:
page_ignore_forwards.patch uploaded by putnopvut (license 60)
Tested by: putnopvut, acunningham
* /, include/asterisk/pbx.h: Merged revisions 164422 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r164422 | mmichelson | 2008-12-15 13:53:08 -0600 (Mon,
15 Dec 2008) | 3 lines Add the deadlock note to
ast_spawn_extension as well ........
* /, include/asterisk/channel.h, include/asterisk/pbx.h: Merged
revisions 164416 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r164416 | mmichelson | 2008-12-15 13:45:07 -0600 (Mon, 15 Dec
2008) | 4 lines Add notes to autoservice and pbx doxygen
regarding a potential deadlock scenario so that it is avoided in
the future ........
2008-12-15 19:48 +0000 [r164417] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c, include/asterisk/strings.h: Revert ast_str
opacity in chan_sip for now, since something wasn't quite right
in the merge.
2008-12-15 19:42 +0000 [r164415] Steve Murphy <murf@digium.com>
* include/asterisk/strings.h: I was getting this warning during a
compile on a 64-bit machine running ubuntu server 8.10, and
gcc-4.3.2: [CXXi] chan_vpb.ii -> chan_vpb.oo cc1plus: warnings
being treated as errors In file included from
/home/murf/asterisk/trunk/include/asterisk/utils.h:671, from
chan_vpb.cc:46:
/home/murf/asterisk/trunk/include/asterisk/strings.h: In function
‘char* ast_str_truncate(ast_str*, ssize_t)’:
/home/murf/asterisk/trunk/include/asterisk/strings.h:479: error:
comparison between signed and unsigned integer expressions
make[1]: *** [chan_vpb.oo] Error 1 make: *** [channels] Error 2
which this fix silences
2008-12-15 18:12 +0000 [r164351] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 164350 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r164350 | file | 2008-12-15 14:11:21 -0400 (Mon, 15 Dec 2008) | 6
lines Do not try to unlock a non-existant channel if the transfer
fails. (closes issue #13800) Reported by: dwagner Patches:
asterisk-1.4.22-chan-sip-nullp.patch uploaded by tweety (license
608) ........
2008-12-15 18:09 +0000 [r164349] Tilghman Lesher <tlesher@digium.com>
* cdr/cdr_pgsql.c: When querying for the structure of the CDR
table, remove the schema, if it exists. (Closes issue #14058)
2008-12-15 17:24 +0000 [r164312] Joshua Colp <jcolp@digium.com>
* main/file.c: Use ast_seekstream to return the file stream back to
the beginning instead of directly seeking to zero. This is
because some audio formats have headers at the front that need to
be skipped, which will be done by the format module. (closes
issue #14079) Reported by: elguero
2008-12-15 17:21 +0000 [r164272-164309] Russell Bryant <russell@digium.com>
* channels/h323/ast_h323.cxx, include/asterisk/strings.h: Fix a
couple more build issues related to ast_str_opaque
* pbx/pbx_dundi.c: When a reload is issued, always process the
configuration for dundi.conf. The reason is that a reload can be
used to refresh DNS lookups for defined peers. Even if the config
file hasn't changed, we want to process it for that purpose.
(closes issue #13776) Reported by: kombjuder
2008-12-15 16:16 +0000 [r164268-164270] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Fix a compile warning and a logic error that
could have been bad for non-realtime queues
* apps/app_queue.c: Fix up a few issues with regards to queues *
Fix reference counting used in the __queues_show function * Add
code to be sure that the "queue show" command does not print
information for a realtime queue which has been deleted from the
backend * Add a missing unref to the realtime queue loading
function for the case where a queue is in the module's container
but has been deleted from the realtime backend (closes issue
#14033) Reported by: cristiandimache Patches: 14033.patch
uploaded by putnopvut (license 60) Tested by: cristiandimache
2008-12-15 15:41 +0000 [r164208-164257] Joshua Colp <jcolp@digium.com>
* configure, include/asterisk/autoconfig.h.in, apps/app_fax.c,
configure.ac: Make app_fax compatible with newer versions of
spandsp. This remains backwards compatible with earlier versions
though so do not fret. (closes issue #14073) Reported by:
seandarcy
* main/utils.c: Update to work with new ast_str changes.
2008-12-15 14:40 +0000 [r164202-164203] Russell Bryant <russell@digium.com>
* main/channel.c, /, main/features.c: Merged revisions 164201 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r164201 | russell | 2008-12-15 08:31:37 -0600 (Mon, 15 Dec 2008)
| 31 lines Handle a case where a call can be bridged to a channel
that is still ringing. The issue that was reported was about a
case where a RINGING channel got redirected to an extension to
pick up a call from parking. Once the parked call got taken out
of parking, it heard silence until the other side answered.
Ideally, the caller that was parked would get a ringing
indication. This patch fixes this case so that the caller
receives ringback once it comes out of parking until the other
side answers. The fixes are: - Make sure we remember that a
channel was an outgoing channel when doing a masquerade. This
prevents an erroneous ast_answer() call on the channel, which
causes a bogus 200 OK to be sent in the case of SIP. - Add some
additional comments to explain related parts of code. - Update
the handling of the ast_channel visible_indication field. Storing
values that are not stateful is pointless. Control frames that
are events or commands should be ignored. - When a bridge first
starts, check to see if the peer channel needs to be given
ringing indication because the calling side is still ringing. -
Rework ast_indicate_data() a bit for the sake of readability.
(closes issue #13747) Reported by: davidw Tested by: russell
Review: http://reviewboard.digium.com/r/90/ ........
* apps/app_jack.c: Fix build WRT ast_str_opaque
2008-12-14 18:16 +0000 [r164168] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/strings.h: Don't pass a negative to an unsigned
type and expect things to work correctly.
2008-12-14 15:26 +0000 [r164054-164137] Sean Bright <sean.bright@gmail.com>
* doc/tex/cdrdriver.tex: Use a \picture instead of ASCII art.
* res/snmp/agent.c: Use ast_str_strlen() instead of recalculating
the string length.
2008-12-13 13:26 +0000 [r164028] Michiel van Baak <michiel@vanbaak.info>
* res/snmp/agent.c: nuke another use of the ast_str internals.
2008-12-13 08:36 +0000 [r163991] Tilghman Lesher <tlesher@digium.com>
* cdr/cdr_sqlite3_custom.c, apps/app_meetme.c,
funcs/func_strings.c, utils/hashtest.c, cdr/cdr_adaptive_odbc.c,
main/utils.c, apps/app_chanisavail.c, include/asterisk/tcptls.h,
cdr/cdr_pgsql.c, res/res_http_post.c, apps/app_followme.c,
res/res_config_sqlite.c, main/config.c, main/cli.c, main/cdr.c,
channels/chan_dahdi.c, res/res_config_odbc.c, main/manager.c,
configure, funcs/func_odbc.c, res/res_agi.c, apps/app_dumpchan.c,
main/logger.c, main/http.c, main/app.c, apps/app_externalivr.c,
res/res_config_ldap.c, include/asterisk/threadstorage.h,
cdr/cdr_manager.c, res/res_clialiases.c, utils/refcounter.c,
res/res_config_pgsql.c, main/strings.c (added), main/pbx.c,
channels/chan_sip.c, main/Makefile, main/translate.c,
include/asterisk/cdr.h, apps/app_queue.c, channels/iax2-parser.c,
funcs/func_realtime.c, utils/Makefile, res/res_config_curl.c,
main/tcptls.c, include/asterisk/app.h, funcs/func_curl.c,
utils/hashtest2.c, include/asterisk/strings.h,
include/asterisk/pbx.h, main/asterisk.c, main/xmldoc.c,
apps/app_voicemail.c, utils/check_expr.c: Merge ast_str_opaque
branch (discontinue usage of ast_str internals)
2008-12-13 03:03 +0000 [r163951-163952] Sean Bright <sean.bright@gmail.com>
* doc/tex/asterisk.tex: This shouldn't have gotten commited. We
might want to generate this into a separate file instead of the
version controlled one.
* doc/tex/qos.tex, doc/tex/asterisk.tex: Use actual tables instead
of ASCII art ones.
2008-12-13 00:59 +0000 [r163912] Joshua Colp <jcolp@digium.com>
* apps/app_chanspy.c: Only detach and destroy the whisper
audiohooks if they are actually in use.
2008-12-12 23:48 +0000 [r163873] Terry Wilson <twilson@digium.com>
* apps/app_queue.c: When using realtime queues, app_queue wasn't
updating the strategy if it was changed in the realtime backend.
This patch resolves the issue for almost all situations. It is
currently not supported to switch to the linear strategy via
realtime since the ao2_container for members will have been set
to have multiple buckets and therefore the members would be
unordered. (closes issue #14034) Reported by: cristiandimache
Tested by: otherwiseguy, cristiandimache
2008-12-12 23:06 +0000 [r163828] Russell Bryant <russell@digium.com>
* res/res_clioriginate.c: Add a note to indicate why this only
supports one channel for now.
2008-12-12 22:04 +0000 [r163762] Tilghman Lesher <tlesher@digium.com>
* main/editline/read.c, /, main/asterisk.c: Merged revisions 163761
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r163761 | tilghman | 2008-12-12 16:03:10 -0600 (Fri, 12 Dec 2008)
| 7 lines Simple fix for Ctrl-C not immediately exiting Asterisk,
but also add a pointer inside editline to look back to
asterisk.c, so others don't spend as much time as I did looking
(in the wrong place) for the appropriate function. Reported by:
ZX81, via the #asterisk-users channel Fixed by: me (license 14)
........
2008-12-12 20:12 +0000 [r163716] Russell Bryant <russell@digium.com>
* CHANGES, res/res_clioriginate.c: Add a new CLI command, "channel
redirect", which is similar in operation to AMI Redirect. Review:
http://reviewboard.digium.com/r/89/
2008-12-12 19:16 +0000 [r163675] Steve Murphy <murf@digium.com>
* channels/chan_dahdi.c: demote always-appearing debug message (for
certain boards) to ast_debug lev 3 msg instead
2008-12-12 18:45 +0000 [r163642-163670] Russell Bryant <russell@digium.com>
* main/tcptls.c, channels/chan_sip.c: Rename a number of
tcptls_session variables. There are no functional changes here.
The name "ser" was used in a lot of places. However, it is a
relic from when the struct was a server_instance, not a
session_instance. It was renamed since it represents both a
server or client connection.
* channels/chan_sip.c: Fix a small race condition in
sip_tcp_locate(). We must increase the reference count on the
tcptls_session _before_ unlocking the thread list.
* channels/chan_sip.c: Resolve crashes when using SIP TCP/TLS with
qualify. The problem was a reference count error on the
tcptls_session structure. (closes issue #13989) Reported by:
Nugget
2008-12-12 18:17 +0000 [r163629] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: When a device registers we need to unlink
them (if linked) from the peers_by_ip container and link them
back in since their IP address has changed. This would have
manifested itself if you configured a new device (as type=peer),
registered, and then tried to place a call from the device. Since
the peer was not linked into the peers_by_ip container it would
have never been found. (closes issue #13811) Reported by: pj
2008-12-12 17:22 +0000 [r163582-163612] Michiel van Baak <michiel@vanbaak.info>
* res/res_monitor.c: Document default Monitor file location.
(closes issue #14065) Reported by: kshumard Patches:
res_monitor.documentation.patch.txt uploaded by kshumard (license
92)
* channels/chan_skinny.c: Fix codec capability setup in chan_skinny
Behaviour now is that general codec config flows to default_line
and default_device. [devices] stuff amends default_device and
similar for [lines]. These are copied to individual device and
line as they are created. Added confcapability and confprefs for
the configured stuff which doesn't change as device and so on are
connected. prefs are based on line prefs if they exist, else the
device prefs are used (prefs identifies codec order). (closes
issue #13806) Reported by: pj Patches: codecs.diff uploaded by
wedhorn (license 30) Tested by: pj and me
2008-12-12 16:55 +0000 [r163579] Joshua Colp <jcolp@digium.com>
* main/channel.c, channels/chan_sip.c: Since chan_sip is callback
devicestate driven do not pass in actual states, pass in unknown
so we get asked. Additionally do not pass in an actual device
state value in ast_setstate since the channel may be callback
driven. (closes issue #13525) Reported by: pj
2008-12-12 15:10 +0000 [r163516] Doug Bailey <dbailey@digium.com>
* configs/phoneprov.conf.sample: Add internationalization to sample
configuration file
2008-12-12 14:44 +0000 [r163449-163512] Russell Bryant <russell@digium.com>
* /, pbx/pbx_dundi.c: Merged revisions 163511 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r163511 | russell | 2008-12-12 08:40:31 -0600 (Fri, 12 Dec 2008)
| 5 lines Specify uint32_t for variables storing a CRC32 so that
it is actually 32 bits on 64-bit machines, as well. (inspired by
issue #13879) ........
* main/channel.c, main/autoservice.c, /,
include/asterisk/channel.h: Merged revisions 163448 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r163448 | russell | 2008-12-12 07:44:08 -0600 (Fri, 12
Dec 2008) | 26 lines Resolve issues that could cause DTMF to be
processed out of order. These changes come from
team/russell/issue_12658 1) Change autoservice to put digits on
the head of the channel's frame readq instead of the tail. If
there were frames on the readq that autoservice had not yet read,
the previous code would have resulted in out of order processing.
This required a new API call to queue a frame to the head of the
queue instead of the tail. 2) Change up the processing of DTMF in
ast_read(). Some of the problems were the result of having two
sources of pending DTMF frames. There was the dtmfq and the more
generic readq. Both were used for pending DTMF in various
scenarios. Simplifying things to only use the frame readq avoids
some of the problems. 3) Fix a bug where a DTMF END frame could
get passed through when it shouldn't have. If code set
END_DTMF_ONLY in the middle of digit emulation, and a digit
arrived before emulation was complete, digits would get processed
out of order. (closes issue #12658) Reported by: dimas Tested by:
russell, file Review: http://reviewboard.digium.com/r/85/
........
2008-12-11 23:38 +0000 [r163384] Tilghman Lesher <tlesher@digium.com>
* /, main/asterisk.c: Merged revisions 163383 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r163383 | tilghman | 2008-12-11 17:35:55 -0600 (Thu, 11 Dec 2008)
| 9 lines When a Ctrl-C or Ctrl-D ends a remote console, on
certain shells, the terminal is messed up. By intercepting those
events with a signal handler in the remote console, we can avoid
those issues. (closes issue #13464) Reported by: tzafrir Patches:
20081110__bug13464.diff.txt uploaded by Corydon76 (license 14)
Tested by: blitzrage ........
2008-12-11 22:49 +0000 [r163317] Matthew Nicholson <mnicholson@digium.com>
* /, pbx/pbx_dundi.c: Merged revisions 163316 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r163316 | mnicholson | 2008-12-11 16:44:31 -0600 (Thu, 11 Dec
2008) | 9 lines Clean up the dundi cache every 5 minutes. (closes
issue #13819) Reported by: adomjan Patches:
pbx_dundi.c-clearcache.patch uploaded by adomjan (license 487)
dundi_clearecache3.diff uploaded by mnicholson (license 96)
Tested by: adomjan ........
2008-12-11 21:48 +0000 [r163241-163254] Russell Bryant <russell@digium.com>
* /, funcs/func_strings.c, funcs/func_cut.c: Merged revisions
163253 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r163253 | russell | 2008-12-11 15:46:29 -0600 (Thu, 11 Dec 2008)
| 8 lines Fix some observed slowdowns in dialplan processing. The
change is to remove autoservice usage from dialplan functions
that do not need it because they do not perform operations that
potentially block. (closes issue #13940) Reported by: tbelder
........
* res/res_timing_pthread.c: Fix a problem where continuous mode
will get inadvertently get turned off if set_rate() is used while
continuous mode was already turned on. (closes issue #13738)
Reported by: smurfix Patches: res.patch.fixed uploaded by smurfix
(license 547)
2008-12-11 20:57 +0000 [r163198-163213] Mark Michelson <mmichelson@digium.com>
* configs/voicemail.conf.sample, apps/app_voicemail.c: Add an
option to voicemail.conf to allow urgent messages to be forwarded
as not urgent. (closes issue #14063) Reported by: jaroth Patches:
urgfwd_v2.patch uploaded by jaroth (license 50)
* main/features.c: Add an appropriate goto if ast_call fails
2008-12-11 20:07 +0000 [r163171] Russell Bryant <russell@digium.com>
* main/channel.c: Fix the "failed" extension for outgoing calls.
The conversion to use ast_check_hangup() everywhere instead of
checking the softhangup flag directly introduced this problem.
The issue is that ast_check_hangup() checked for tech_pvt to be
NULL. Unfortunately, this will be NULL is some valid
circumstances, such as with a dummy channel. The fix is simple.
Don't check tech_pvt. It's pointless, because the code path that
sets this to NULL is when the channel hangup callback gets
called. This happens inside of ast_hangup(), which is the same
function responsible for freeing the channel. Any code calling
ast_check_hangup() better not be calling it after that point, and
if so, we have a bigger problem at hand. (closes issue #14035)
Reported by: erogoza
2008-12-11 20:02 +0000 [r163168] Tilghman Lesher <tlesher@digium.com>
* configure, configure.ac: Sometimes even Linux needs -lm to link
libtonezone, such as when libtonezone is compiled statically.
(closes issue #13887) Reported by: tzafrir
2008-12-11 19:40 +0000 [r163166] Mark Michelson <mmichelson@digium.com>
* main/features.c: Reduce indentation level of
ast_feature_request_and_dial
2008-12-11 17:06 +0000 [r163094] Russell Bryant <russell@digium.com>
* /, main/features.c: Merged revisions 163092 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r163092 | russell | 2008-12-11 10:54:51 -0600 (Thu, 11 Dec 2008)
| 11 lines Fix an issue that made it so you could only have a
single caller executing a custom feature at a time. This was
especially problematic when custom features ran for any
appreciable amount of time. The fix turned out to be quite
simple. The dynamic features are now stored in a read/write list
instead of a list using a mutex. (closes issue #13478) Reported
by: neutrino88 Fix suggested by file ........
2008-12-11 16:52 +0000 [r163089] Tilghman Lesher <tlesher@digium.com>
* /, res/res_agi.c: Merged revisions 163088 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r163088 | tilghman | 2008-12-11 10:51:27 -0600 (Thu, 11 Dec 2008)
| 6 lines Don't wait forever, if there's a specified recording
timeout. (closes issue #13885) Reported by: bamby Patches:
res_agi.c.patch uploaded by bamby (license 430) ........
2008-12-11 16:47 +0000 [r163081-163085] Mark Michelson <mmichelson@digium.com>
* /, apps/app_queue.c: Merged revisions 163084 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r163084 | mmichelson | 2008-12-11 10:46:22 -0600 (Thu, 11 Dec
2008) | 4 lines Revert this cast to long. Using time_t here
causes build failures on a FreeBSD 32-bit build. ........
* /, apps/app_queue.c: Merged revisions 163080 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r163080 | mmichelson | 2008-12-11 10:24:43 -0600 (Thu, 11 Dec
2008) | 14 lines Fix a potential crash due to unsafe datastore
handling. This patch also contains a conversion from using long
to time_t for representing times for a queue, as well as some
whitespace fixes. (closes issue #14060) Reported by: nivek
Patches: datastore_fixup.patch.corrected uploaded by nivek
(license 636) with slight modification from me Tested by: nivek
........
2008-12-11 15:40 +0000 [r163037] Sean Bright <sean.bright@gmail.com>
* doc/tex/qos.tex: Fix some of the grammar issues in
doc/tex/qos.tex. (closes issue #14049) Reported by: kshumard
Patches: doc.tex.qos.tex.patch uploaded by kshumard (license 92)
(Slight modifications by seanbright)
2008-12-11 15:05 +0000 [r162997] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: When a device registers to use it is
entirely possible that they may be in use, so tell the core that
we don't know the devstate and have it ask us for it. (closes
issue #13525) Reported by: pj
2008-12-10 23:01 +0000 [r162930] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c: Previously missing line, now the substitution works
correctly
2008-12-10 22:53 +0000 [r162927] Jeff Peeler <jpeeler@digium.com>
* /, res/res_musiconhold.c: Merged revisions 162926 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r162926 | jpeeler | 2008-12-10 16:52:51 -0600 (Wed, 10
Dec 2008) | 3 lines Oops, inverted logic for a strcasecmp check.
Pointed out by mmichelson, thanks! ........
2008-12-10 22:48 +0000 [r162923] Joshua Colp <jcolp@digium.com>
* res/res_clialiases.c: Fix reloads of aliased CLI commands. Due to
changes done to turn it into a single memory allocation we can't
just use the existing CLI alias structure. We have to destroy all
existing ones and then create new ones. (closes issue #14054)
Reported by: pj
2008-12-10 22:48 +0000 [r162922] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c: Checking global variables here actually overwrote the
previous substitution by channel variables, and in any case, was
redundant; pbx_substitute_variables_helper ALREADY does
substitution for global variables. (closes issue #13327) Reported
by: pj
2008-12-10 22:11 +0000 [r162891] Jeff Peeler <jpeeler@digium.com>
* /, res/res_musiconhold.c: Merged revisions 162874 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r162874 | jpeeler | 2008-12-10 16:04:18 -0600 (Wed, 10
Dec 2008) | 5 lines (closes issue #13229) Reported by:
clegall_proformatique Ensure that moh_generate does not return
prematurely before local_ast_moh_stop is called. Also, the sleep
in mp3_spawn now only occurs for http locations since it seems to
have been added originally only for failing media streams.
........
2008-12-10 19:02 +0000 [r162739-162805] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 162804 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r162804 | file | 2008-12-10 15:01:17 -0400 (Wed, 10 Dec 2008) | 6
lines Fix subscription based MWI up a bit. We only want to put
sip: at the beginning of the URI if it is not already there and
revert code to ignore destination check if subscribing for MWI.
(closes issue #12560) Reported by: vsauer Patches: patch001.diff
uploaded by ramonpeek (license 266) ........
* /, channels/chan_sip.c: Merged revisions 162738 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r162738 | file | 2008-12-10 13:50:43 -0400 (Wed, 10 Dec 2008) | 6
lines When a SIP peer unregisters set the expiry time back to 0
so that the 200 OK contains an expires of 0. (closes issue
#13599) Reported by: hjourdain Patches: chan_sip.c.diff uploaded
by hjourdain (license 583) ........
2008-12-10 17:09 +0000 [r162687] Michiel van Baak <michiel@vanbaak.info>
* include/asterisk.h, main/asterisk.c, main/cli.c: add tab
completion for 'core set debug X filename.c' (closes issue
#13969) Reported by: jtodd Patches: 20081205__bug13969.diff.txt
uploaded by Corydon76 (license 14) Tested by: mvanbaak, eliel
2008-12-10 16:39 +0000 [r162664-162667] Mark Michelson <mmichelson@digium.com>
* doc/tex/misdn.tex, /: Merged revisions 162659 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r162659 | mmichelson | 2008-12-10 10:10:25 -0600 (Wed, 10 Dec
2008) | 8 lines Add missing documentation to misdn.txt (closes
issue #14052) Reported by: festr Patches: misdn.txt.patch
uploaded by festr (license 443) ........
* /, channels/chan_sip.c: Merged revisions 162663 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r162663 | mmichelson | 2008-12-10 10:24:56 -0600 (Wed, 10 Dec
2008) | 11 lines Revert fix for issue 13570. It has caused more
problems than it helped to fix. (closes issue #13783) Reported
by: navkumar (closes issue #14025) Reported by: ffs ........
2008-12-10 16:11 +0000 [r162619-162660] Joshua Colp <jcolp@digium.com>
* res/res_http_post.c: FreeBSD also needs libgen.h (closes issue
#14051) Reported by: ys Patches: res_http_post.c.diff uploaded by
ys (license 281)
* /, main/rtp.c: Merged revisions 162653 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r162653 | file | 2008-12-10 12:05:29 -0400 (Wed, 10 Dec 2008) | 6
lines Increment the sequence number on the end packets for
RFC2833. After reading the RFC some more and doing some testing I
agree with this change. (closes issue #12983) Reported by: vt
Patches: dtmf_inc_seqnum_on_end_pkts.diff uploaded by vt (license
520) ........
* channels/chan_sip.c: When transmitting a register set the socket
port to the local one for the transport being used, not the port
for the remote server. (closes issue #13633) Reported by:
performer
2008-12-10 11:34 +0000 [r162583] Michiel van Baak <michiel@vanbaak.info>
* res/snmp/agent.c: Make res_snmp.so compile on OpenBSD. OpenBSD
uses an old version of gcc which throws an error if you use a
macro that's not #defined
2008-12-10 01:09 +0000 [r162542] Joshua Colp <jcolp@digium.com>
* doc/janitor-projects.txt, channels/iax2-parser.c,
apps/app_voicemail.c: Finish conversion to using ARRAY_LEN and
remove it as a janitor project. (closes issue #14032) Reported
by: bkruse Patches: 14032.patch uploaded by bkruse (license 132)
2008-12-09 23:41 +0000 [r162488] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk/stringfields.h: it does help if the compiler
attribute syntax is correct
2008-12-09 23:10 +0000 [r162466] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_voicemail.c: Merged revisions 162463 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r162463 | tilghman | 2008-12-09 17:08:53 -0600 (Tue, 09
Dec 2008) | 2 lines Oops, should be "tz", not "zonetag". ........
2008-12-09 22:38 +0000 [r162414-162418] Russell Bryant <russell@digium.com>
* include/asterisk/doxyref.h, contrib/asterisk-ng-doxygen,
main/asterisk.c: Add some additional Asterisk project developer
documentation. After the nightly update of the documentation on
asterisk.org, I'll post an update to asterisk-dev with a pointer
to the changes. This covers some release branch and commit policy
information. None of this should be a surprise, since it's just
documenting what we have already been doing.
* include/asterisk/utils.h, /, main/utils.c, main/asterisk.c:
Merged revisions 162413 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r162413 | russell | 2008-12-09 16:17:39 -0600 (Tue, 09 Dec 2008)
| 8 lines Remove the test_for_thread_safety() function
completely. The test is not valid. Besides, if we actually
suspected that recursive mutexes were not working, we would get a
ton of LOG_ERROR messages when DEBUG_THREADS is turned on.
(inspired by a discussion on the asterisk-dev list) ........
2008-12-09 21:57 +0000 [r162355] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_voicemail.c: Merged revisions 162348 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r162348 | tilghman | 2008-12-09 15:53:25 -0600 (Tue, 09
Dec 2008) | 4 lines We appear to have documented tz= in the
[general] section of voicemail.conf, without actually having
implemented it. Oops. (Reported by Olivier on the -users list)
........
2008-12-09 21:16 +0000 [r162342] Joshua Colp <jcolp@digium.com>
* /, apps/app_directed_pickup.c: Merged revisions 162341 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r162341 | file | 2008-12-09 17:14:29 -0400 (Tue, 09 Dec 2008) | 4
lines Add 'down' as a valid state for directed call pickup. This
creeps up when we receive session progress when dialing a device
and not ringing. (closes issue #14005) Reported by: ddl ........
2008-12-09 20:59 +0000 [r162291] Russell Bryant <russell@digium.com>
* /, apps/app_meetme.c: Merged revisions 162286 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r162286 | russell | 2008-12-09 14:57:35 -0600 (Tue, 09 Dec 2008)
| 9 lines Fix an issue where callers on an incoming call on an
SLA trunk would not hear ringback. We need to make sure that we
don't start writing audio to the trunk channel until we're
actually ready to answer it. Otherwise, the channel driver will
treat it as inband progress, even though all they are getting is
silence. (closes issue #12471) Reported by: mthomasslo ........
2008-12-09 20:46 +0000 [r162275] Joshua Colp <jcolp@digium.com>
* /, apps/app_festival.c: Merged revisions 162273 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r162273 | file | 2008-12-09 16:44:32 -0400 (Tue, 09 Dec 2008) | 4
lines Fix double declaration of 'x' on the PPC platform. (closes
issue #14038) Reported by: ffloimair ........
2008-12-09 20:40 +0000 [r162271] Steve Murphy <murf@digium.com>
* /, res/ael/ael_lex.c, res/ael/ael.flex: Merged revisions 162264
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r162264 | murf | 2008-12-09 13:20:54 -0700 (Tue, 09 Dec 2008) | 1
line In discussion with seanbright on #asterisk-dev, I have added
a default rule, and an option to suppress the default rule from
being generated in the flex output, for the sake of those OS's
where they didn't tweak flex's ECHO macro, and the compiler
doesn't like it. The regressions are OK with this. ........
2008-12-09 20:30 +0000 [r162266] Mark Michelson <mmichelson@digium.com>
* main/pbx.c, /: Merged revisions 162265 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r162265 | mmichelson | 2008-12-09 14:28:44 -0600 (Tue, 09 Dec
2008) | 6 lines If we fail to start a thread for the pbx to run
in, we need to be sure to decrease the number of active calls on
the system. This fix may relate to ABE-1713, but it is not
certain yet. ........
2008-12-09 19:48 +0000 [r162197-162205] Joshua Colp <jcolp@digium.com>
* /, main/rtp.c: Merged revisions 162204 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r162204 | file | 2008-12-09 15:47:07 -0400 (Tue, 09 Dec 2008) | 7
lines Make sure that the timestamp for DTMF is not the same as
the previous voice frame and do not send audio when transmitting
DTMF as this confuses some equipment. (closes issue #13209)
Reported by: ip-rob Patches: 13209.diff uploaded by file (license
11) Tested by: ip-rob, bujones ........
* /, main/rtp.c: Merged revisions 162188 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r162188 | file | 2008-12-09 15:06:14 -0400 (Tue, 09 Dec 2008) | 4
lines Take video into account when early bridging RTP. (closes
issue #13535) Reported by: davidw ........
2008-12-09 18:35 +0000 [r162079-162140] Steve Murphy <murf@digium.com>
* /, res/ael/ael_lex.c, res/ael/ael.flex: Merged revisions 162136
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r162136 | murf | 2008-12-09 11:13:39 -0700 (Tue, 09 Dec 2008) | 1
line Previous fix used ast_malloc and ast_copy_string and messed
up the standalone stuff. Fixed. ........
* res/ael/pval.c, /, include/asterisk/pval.h, res/ael/ael_lex.c,
res/ael/ael.flex: Merged revisions 162013 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r162013 | murf | 2008-12-09 09:31:55 -0700 (Tue, 09 Dec 2008) |
45 lines (closes issue #14019) Reported by: ckjohnsonme Patches:
14019.diff uploaded by murf (license 17) Tested by: ckjohnsonme,
murf This crash was the result of a few small errors that would
combine in 64-bit land to result in a crash. 32-bit land might
have seen these combine to mysteriously drop the args to an
application call, in certain circumstances. Also, in trying to
find this bug, I spotted a situation in the flex input, where, in
passing back a 'word' to the parser, it would allocate a buffer
larger than necessary. I changed the usage in such situations, so
that strdup was not used, but rather, an ast_malloc, followed by
ast_copy_string. I removed a field from the pval struct, in u2,
that was never getting used, and set in one spot in the code. I
believe it was an artifact of a previous fix to make switch cases
work invisibly with extens. And, for goto's I removed a '!' from
before a strcmp, that has been there since the initial merging of
AEL2, that might prevent the proper target of a goto from being
found. This was pretty harmless on its own, as it would just
louse up a consistency check for users. Many thanks to
ckjohnsonme for providing a simplified and complete set of
information about the bug, that helped considerably in finding
and fixing the problem. Now, to get aelparse up and running again
in trunk, and out of its "horribly broken" state, so I can run
the regression suite! ........
2008-12-09 16:47 +0000 [r161951-162016] Russell Bryant <russell@digium.com>
* /, apps/app_disa.c: Merged revisions 162014 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r162014 | russell | 2008-12-09 10:46:53 -0600 (Tue, 09 Dec 2008)
| 5 lines Allow DISA to handle extensions that start with #.
(closes issue #13330) Reported by: jcovert ........
* /, main/app.c: Merged revisions 161948 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r161948 | russell | 2008-12-09 08:52:25 -0600 (Tue, 09 Dec 2008)
| 15 lines Fix a problem with GROUP() settings on a masquerade.
The previous code carried over group settings from the old
channel to the new one. However, it did nothing with the group
settings that were already on the new channel. This patch removes
all group settings that already existed on the new channel. I
have a more complicated version of this patch which addresses
only the most blatant problem with this, which is that a channel
can end up with multiple group settings in the same category.
However, I could not think of a use case for keeping any of the
group settings from the old channel, so I went this route for
now. (closes AST-152) ........
2008-12-09 14:49 +0000 [r161947] Eliel C. Sardanons <eliels@gmail.com>
* funcs/func_odbc.c: Avoid allocating memory for a thread that
don't need it. Also, this memory was not being freed until the
main thread ends. (That is never). (closes issue #14040) Reported
by: eliel Patches: func_odbc.c.patch uploaded by eliel (license
64)
2008-12-08 23:04 +0000 [r161911] Brandon Kruse <bkruse@digium.com>
* main/pbx.c: Note that the recently changed waittime parameter is
in milliseconds.
2008-12-08 21:41 +0000 [r161830-161869] Joshua Colp <jcolp@digium.com>
* formats/format_pcm.c: Add alw as a valid file extension for alaw
and ulw as a valid file extension for ulaw. (closes issue #14001)
Reported by: henrikw Patches: alw.diff uploaded by henrikw
(license 627)
* contrib/scripts/autosupport.8, contrib/scripts/autosupport:
Update autosupport script with a few changes.
2008-12-08 18:49 +0000 [r161790] Tilghman Lesher <tlesher@digium.com>
* main/manager.c: Allocate enough space initially for the message.
(closes issue #14027) Reported by: junky Patches: M14027.diff
uploaded by junky (license 177)
2008-12-08 18:47 +0000 [r161726-161787] Joshua Colp <jcolp@digium.com>
* main/pbx.c: Fix a regression introduced when the PBX timeouts
were converted to milliseconds. collect_digits now gets
milliseconds fed to it, not seconds. (closes issue #14012)
Reported by: dveiga Patches: 14012.patch uploaded by bkruse
(license 132)
* /, channels/chan_sip.c: Merged revisions 161725 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r161725 | file | 2008-12-08 13:52:10 -0400 (Mon, 08 Dec 2008) | 6
lines Make the usereqphone option work again. (closes issue
#13474) Reported by: mmaguire Patches: 20080912_bug13474.diff
uploaded by mmaguire (license 571) ........
2008-12-08 17:23 +0000 [r161721] Matthew Nicholson <mnicholson@digium.com>
* channels/chan_sip.c: Fix a crash that can occur on a transfer in
chan_sip when attempting to collect rtp stats. (closes issue
#13956) Reported by: chris-mac Tested by: chris-mac
2008-12-08 16:02 +0000 [r161679] Terry Wilson <twilson@digium.com>
* channels/chan_sip.c, CHANGES: Add the ability to play a courtesy
tone to the transfer target in a native SIP attended transfer by
setting the variable ATTENEDED_TRANSFER_COMPLETE_SOUND.
2008-12-08 04:23 +0000 [r161571-161637] Eliel C. Sardanons <eliels@gmail.com>
* main/xmldoc.c: - Fix a leak while printing an argument
description. - Avoid printing the name of an argument in the
[Arguments] tag if there is no description for that argument.
* apps/app_voicemail.c: Add voicemail related applications and
functions XML documentation: applications: - VoiceMail() -
VoiceMailMain() - MailboxExists() - VMAuthenticate() functions: -
MAILBOX_EXISTS()
* apps/app_sms.c: Introduce SMS() application XML documentation.
2008-12-06 21:18 +0000 [r161536] Eliel C. Sardanons <eliels@gmail.com>
* apps/app_speech_utils.c: Move Speech* applications and functions
documentation to XML.
2008-12-05 23:24 +0000 [r161493] Mark Michelson <mmichelson@digium.com>
* apps/app_stack.c: If the autoloop flag is set on a channel, then
we need to add 1 to the priority when checking if the extension
exists. Otherwise, gosubs will fail. This was discovered when
investigating an asterisk-users mailing list post made by Gary
Hawkins.
2008-12-05 21:08 +0000 [r161349-161427] Sean Bright <sean.bright@gmail.com>
* /, include/asterisk/astobj2.h, main/astobj2.c: Merged revisions
161426 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
................ r161426 | seanbright | 2008-12-05 16:02:20 -0500
(Fri, 05 Dec 2008) | 15 lines Merged revisions 161421 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r161421 | seanbright | 2008-12-05 15:50:23 -0500 (Fri, 05 Dec
2008) | 8 lines Fix build errors on FreeBSD (uint -> unsigned
int). (closes issue #14006) Reported by: alphaque Patches:
astobj2.h-patch uploaded by alphaque (license 259) (Slightly
modified by seanbright) ........ ................
* apps/app_voicemail.c: Use ast_free() instead of free(), pointed
out by eliel on IRC.
* apps/app_voicemail.c: When using IMAP_STORAGE, it's important to
convert bare newlines (\n) in emailbody and pagerbody to CR-LF so
that the IMAP server doesn't spit out an error. This was
informally reported on #asterisk-dev a few weeks ago. Reviewed by
Mark M. on IRC.
2008-12-05 14:16 +0000 [r161252-161288] Russell Bryant <russell@digium.com>
* main/pbx.c, /: Merged revisions 161287 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r161287 | russell | 2008-12-05 08:12:14 -0600 (Fri, 05 Dec 2008)
| 2 lines Fix a NULL format string warning found by buildbot.
........
* apps/app_minivm.c: Resolve a compiler warning from buildbot about
a NULL format string.
2008-12-05 10:31 +0000 [r161218] Eliel C. Sardanons <eliels@gmail.com>
* main/udptl.c, main/frame.c, res/res_musiconhold.c,
channels/chan_iax2.c, res/res_jabber.c, res/res_config_sqlite.c,
main/config.c, main/cli.c, channels/chan_dahdi.c, main/manager.c,
channels/chan_skinny.c, res/res_agi.c, main/features.c,
apps/app_minivm.c, pbx/pbx_ael.c, main/logger.c, main/http.c,
res/res_realtime.c, channels/chan_alsa.c, res/res_config_ldap.c,
apps/app_rpt.c, main/db.c, res/res_config_pgsql.c, main/pbx.c,
channels/chan_sip.c, main/translate.c, channels/chan_agent.c,
res/res_convert.c, res/res_crypto.c, apps/app_queue.c,
channels/chan_oss.c, apps/app_playback.c,
channels/chan_usbradio.c, main/file.c, main/astmm.c,
pbx/pbx_dundi.c, res/res_indications.c, pbx/pbx_config.c,
apps/app_mixmonitor.c, res/res_odbc.c, main/asterisk.c,
apps/app_voicemail.c: Janitor, use ARRAY_LEN() when possible.
(closes issue #13990) Reported by: eliel Patches: array_len.diff
uploaded by eliel (license 64)
2008-12-05 05:41 +0000 [r161181] Tilghman Lesher <tlesher@digium.com>
* main/config.c: The first file should have a blank config filename
in the structure, so that when a save occurs to a different
filename, everything goes to the alternate filename, instead of
appending to the original. This is important for the AMI command
UpdateConfig. (closes issue #13301) Reported by: trevo Patches:
20081113__bug13301.diff.txt uploaded by Corydon76 (license 14)
20081113__bug13301__1.6.0.diff.txt uploaded by Corydon76 (license
14) Tested by: Corydon76, blitzrage
2008-12-05 02:47 +0000 [r161147] Sean Bright <sean.bright@gmail.com>
* apps/app_voicemail.c: Check the return value of fread/fwrite so
the compiler doesn't complain. Only a problem when IMAP_STORAGE
is enabled.
2008-12-04 23:00 +0000 [r161115] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: If
'faxdetect=yes' in sip.conf, switch to a 'fax' extension (if it
exists) after T38 is negotiated. Terry Wilson created the
original patch for this functionality, which I slightly modified
and added the faxdetect=yes|no configuration option. This patch
is only for T38 fax detection and does not do anything for G711
over SIP fax detection. By default, this option is disabled.
Reviewboard: http://reviewboard.digium.com/r/69/ This
functionality is for issue AST-140.
2008-12-04 19:31 +0000 [r161077] Eliel C. Sardanons <eliels@gmail.com>
* main/cli.c: Fix minor coding guidelines introduced with CLI
permissions.
2008-12-04 18:32 +0000 [r161014] Jeff Peeler <jpeeler@digium.com>
* /, main/rtp.c: Merged revisions 161013 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r161013 | jpeeler | 2008-12-04 12:30:41 -0600 (Thu, 04 Dec 2008)
| 9 lines (closes issue #13835) Reported by: matt_b Tested by:
jpeeler This mirrors a check that was present in ast_rtp_read to
also be in ast_rtp_raw_write to not schedule sending the receiver
report if the remote RTCP endpoint address isn't present in the
RTCP structure. Closes AST-142. ........
2008-12-04 16:45 +0000 [r160945] Mark Michelson <mmichelson@digium.com>
* /, main/callerid.c: Merged revisions 160943 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r160943 | mmichelson | 2008-12-04 10:44:18 -0600 (Thu, 04 Dec
2008) | 15 lines Fix a callerid parsing issue. If someone
formatted callerid like the following: "name <number>" (including
the quotation marks), then the parts would be parsed as name:
"name number: number This is because the closing quotation mark
was not discovered since the number and everything after was
parsed out of the string earlier. Now, there is a check to see if
the closing quote occurs after the number, so that we can know if
we should strip off the opening quote on the name. Closes AST-158
........
2008-12-04 16:37 +0000 [r160938] Michiel van Baak <michiel@vanbaak.info>
* build_tools/cflags-devmode.xml, channels/chan_skinny.c: Add debug
flag so skinny debug will show information about packets. We dont
want to scare users with this, so we added a devmode compile flag
(closes issue #13952) Reported by: wedhorn Patches:
packetdebug3.diff uploaded by wedhorn (license 30) Tested by:
mvanbaak, wedhorn
2008-12-04 13:45 +0000 [r160896] Eliel C. Sardanons <eliels@gmail.com>
* res/res_agi.c: Added XML documentation for the following AGI
commands: - get option - get variable - hangup - noop
2008-12-04 01:36 +0000 [r160854-160856] Richard Mudgett <rmudgett@digium.com>
* funcs/func_callerid.c: Jcolp pointed out that num will also match
number
* funcs/func_callerid.c: * Found a couple more places where
num/number needed to be done so 1.4 upgraders will not have
problems. * Added curly braces and minor tweaks.
2008-12-03 21:58 +0000 [r160791] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_voicemail.c: Merged revisions 160770 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r160770 | tilghman | 2008-12-03 15:54:07 -0600 (Wed, 03
Dec 2008) | 2 lines Some compilers warn on null format strings;
some don't (caught by buildbot) ........
2008-12-03 21:09 +0000 [r160760] Steve Murphy <murf@digium.com>
* /, funcs/func_callerid.c: Merged revisions 160703 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r160703 | murf | 2008-12-03 13:41:42 -0700 (Wed, 03 Dec
2008) | 11 lines (closes issue #13597) Reported by: john8675309
Patches: patch.13597 uploaded by murf (license 17) Tested by:
murf, john8675309 This patch causes the setcid func to update the
CDR clid after setting the channel field. I also notice that in
trunk, the num/number of 1.4 is left out; I decided to include
the option to use either in trunk, so as not to have 1.4
upgraders not to have problems. ........
2008-12-03 20:35 +0000 [r160699-160700] Jason Parker <jparker@digium.com>
* main/manager.c: Another place this is missing
* main/manager.c: Fix typo when ListCategories returns none.
(closes issue #13994) Reported by: mika Patches:
ListCategoriesActionPatch.diff uploaded by mika (license 624)
2008-12-03 19:25 +0000 [r160663] Eliel C. Sardanons <eliels@gmail.com>
* channels/iax2-provision.c: - iax2-provision was not freeing
iax_templates structure when unloading the chan_iax2.so module. -
Move the code to start using the LIST macros. Review:
http://reviewboard.digium.com/r/72 (closes issue #13232) Reported
by: eliel Patches: iax2-provision.patch.txt uploaded by eliel
(license 64) (with minor changes pointed by Mark Michelson on
review board) Tested by: eliel
2008-12-03 18:37 +0000 [r160626] Mark Michelson <mmichelson@digium.com>
* apps/app_dial.c, apps/app_queue.c, apps/app_stack.c: Add some
safety measures when using gosub, especially when using the
options for app_dial and app_queue to run a gosub when the call
is answered. * Check for the existence of the gosub target in
gosub_exec. If it is nonexistent, then this will cause errors
when we attempt to actually run the gosub, including a definite
memory leak and potential crashes. Return an error in this
situation * Check the return value of pbx_exec in app_dial and
app_queue before attempting to actually run the gosub routine. If
there was an error, we should not attempt to run the gosub. *
Change a '|' to a ',' in app_queue. * Add some extra curly braces
where they had been missing previously. (closes issue #13548)
Reported by: fiddur
2008-12-03 17:48 +0000 [r160562] Eliel C. Sardanons <eliels@gmail.com>
* apps/app_minivm.c: - Add <variable /> tags when naming a channel
variable. - Add <filename /> tags when naming a filename. -
Simplify the xml formatting putting some enters.
2008-12-03 17:38 +0000 [r160559] Tilghman Lesher <tlesher@digium.com>
* pbx/pbx_spool.c, /: Merged revisions 160558 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r160558 | tilghman | 2008-12-03 11:34:34 -0600 (Wed, 03 Dec 2008)
| 7 lines If an entry is added to the directory during a scan
when another entry expires, then that new entry will not be
processed promptly, but must wait for either a future entry to
start or a current entry's retry to occur. If no other entries
exist in the directory (other than the new entries) when a bunch
expire, then the new entries must wait until another new entry is
added to be processed. This was a rather weird race condition,
really. Fixes AST-147. ........
2008-12-03 17:07 +0000 [r160555] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: When investigating issue #13548, I found that
gosub handling in app_queue was just completely wrong, mostly
because the channel operations being performed were being done on
the incorrect channel. With this set of changes, a gosub will
correctly run on the answering queue member's channel. There are
still crash issues which occur if there are dialplan syntax
errors, so I cannot yet close the referenced issue.
2008-12-03 17:01 +0000 [r160481-160552] Tilghman Lesher <tlesher@digium.com>
* pbx/pbx_spool.c, /: Merged revisions 160551 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r160551 | tilghman | 2008-12-03 10:58:34 -0600 (Wed, 03 Dec 2008)
| 4 lines Don't start scanning the directory until all modules
are loaded, because some required modules (channels, apps,
functions) may not yet be in memory yet. Fixes AST-149. ........
* /, channels/chan_sip.c: Merged revisions 160480 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r160480 | tilghman | 2008-12-03 08:09:35 -0600 (Wed, 03 Dec 2008)
| 7 lines Jon Bonilla (Manwe) pointed out on the -dev list: "I
guess that having only ip-phones in mind is not a good approach.
Since it is possible to have a sip proxy connected to asterisk we
could receive a 407 (unauthorized) or 483 (too many hops) as
response and dialog ending would not be a good behavior." So
modified. ........
2008-12-03 11:01 +0000 [r160447] Eliel C. Sardanons <eliels@gmail.com>
* apps/app_stack.c: - Avoid setting .synopsis and .syntax if we are
using XML documentation (or the xml documentation wont be
loaded). - Use <variable></variable> to refer to a dialplan
variable.
2008-12-02 18:48 +0000 [r160344-160346] Tilghman Lesher <tlesher@digium.com>
* CHANGES: Info on LOCAL_PEEK function.
* apps/app_stack.c: Add LOCAL_PEEK function, as requested by
lmadsen.
2008-12-02 18:04 +0000 [r160319-160333] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c: remove duplicate comment that I
accidentally merged
* channels/chan_dahdi.c: (closes issue #13786) Reported by: tzafrir
Readding DAHDI_CHECK_HOOKSTATE define that was removed in r134260
which fixes not being able to make outgoing calls on some FXO
adapters:
http://lists.digium.com/pipermail/asterisk-users/2008-November/thread.html#221553
2008-12-02 17:56 +0000 [r160208-160308] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_sip.c: Merged revisions 160297 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r160297 | tilghman | 2008-12-02 11:42:09 -0600 (Tue, 02 Dec 2008)
| 10 lines When the text does not match exactly (e.g. RTP/SAVP),
then the %n conversion fails, and the resulting integer is
garbage. Thus, we must initialize the integer and check it
afterwards for success. (closes issue #14000) Reported by: folke
Patches: asterisk-sipbg-sscanf-1.4.22.diff uploaded by folke
(license 626) asterisk-sipbg-sscanf-1.6.0.1.diff uploaded by
folke (license 626) asterisk-sipbg-sscanf-trunk-r159896.diff
uploaded by folke (license 626) ........
* main/pbx.c, main/frame.c, /, channels/chan_features.c,
include/asterisk/stringfields.h, apps/app_voicemail.c,
main/cli.c: Merged revisions 160207 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r160207 | tilghman | 2008-12-01 18:25:16 -0600 (Mon, 01 Dec 2008)
| 3 lines Ensure that Asterisk builds with --enable-dev-mode,
even on the latest gcc and glibc. ........
2008-12-01 23:37 +0000 [r160170-160172] Sean Bright <sean.bright@gmail.com>
* main/manager.c, /: Merged revisions 159976 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r159976 | mvanbaak | 2008-12-01 11:08:36 -0500 (Mon, 01 Dec 2008)
| 3 lines Get rid of the useless format string and argument in
the Bogus/ manager channelname. Noted by kpfleming and name
Bogus/manager suggested by eliel ........
* channels/chan_phone.c: Silence a build warning.
(chan_phone.c:810: warning: value computed is not used)
* utils/smsq.c: Pay attention to the return value of system(), even
if we basically ignore it.
2008-12-01 21:23 +0000 [r160097] Tilghman Lesher <tlesher@digium.com>
* configure, configure.ac: Use AST_EXT_LIB_SETUP before using
AST_EXT_LIB_CHECK or bad things happen.
2008-12-01 18:52 +0000 [r160062] Eliel C. Sardanons <eliels@gmail.com>
* configs/cli_permissions.conf.sample (added), configure,
include/asterisk/autoconfig.h.in, configure.ac,
include/asterisk/cli.h, include/asterisk/_private.h, CHANGES,
main/asterisk.c, main/cli.c: Introduce CLI permissions. Based on
cli_permissions.conf configuration file, we are able to permit or
deny cli commands based on some patterns and the local user and
group running rasterisk. (Sorry if I missed some of the testers).
Reviewboard: http://reviewboard.digium.com/r/11/ (closes issue
#11123) Reported by: eliel Tested by: eliel, IgorG, Laureano,
otherwiseguy, mvanbaak
2008-12-01 17:34 +0000 [r159911-160004] Russell Bryant <russell@digium.com>
* /, channels/chan_iax2.c: Merged revisions 160003 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r160003 | russell | 2008-12-01 11:27:30 -0600 (Mon, 01
Dec 2008) | 6 lines Apply some logic used in iax2_indicate() to
iax2_setoption(), as well, since they both have the potential to
send control frames in the middle of call setup. We have to wait
until we have received a message back from the remote end before
we try to send any more frames. Otherwise, the remote end will
consider it invalid, and we'll get stuck in an INVAL/VNAK storm.
........
* /, .cleancount: Merged revisions 159900 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r159900 | russell | 2008-12-01 08:52:56 -0600 (Mon, 01 Dec 2008)
| 2 lines Force a "make clean" to avoid a bizarre build issue ...
........
2008-12-01 14:09 +0000 [r159898] Michiel van Baak <michiel@vanbaak.info>
* main/manager.c, /: Merged revisions 159897 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r159897 | mvanbaak | 2008-12-01 15:05:41 +0100 (Mon, 01 Dec 2008)
| 4 lines make manager compile on OpenBSD. The last (10th)
argument to ast_channel_alloc here should be a pointer and NULL
is not really a pointer. ........
2008-11-29 18:33 +0000 [r159853] Tilghman Lesher <tlesher@digium.com>
* apps/app_readexten.c: Allow the '#' sign to exist within an
extension (inspired by issue #13330)
2008-11-29 17:57 +0000 [r159774-159818] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_vpb.cc, /, main/utils.c, channels/chan_iax2.c,
utils/frame.c, include/asterisk/astmm.h, configure,
include/asterisk/compat.h, main/features.c,
include/asterisk/module.h, main/logger.c,
include/asterisk/dlinkedlists.h, main/dns.c,
include/asterisk/utils.h, include/asterisk/devicestate.h,
channels/chan_sip.c, include/asterisk/dundi.h,
include/asterisk/enum.h, configure.ac, channels/chan_agent.c,
include/asterisk/config.h, utils/astman.c,
include/asterisk/cli.h, include/asterisk/channel.h,
include/jitterbuf.h, include/asterisk/manager.h,
utils/conf2ael.c, cdr/cdr_tds.c, main/ast_expr2.c,
include/asterisk/logger.h, Makefile, include/asterisk/res_odbc.h,
main/srv.c, channels/chan_misdn.c,
include/asterisk/linkedlists.h, main/event.c,
include/asterisk/lock.h, include/asterisk/strings.h,
utils/extconf.c, makeopts.in, include/asterisk/stringfields.h,
main/xmldoc.c, utils/check_expr.c: incorporates r159808 from
branches/1.4:
------------------------------------------------------------------------
r159808 | kpfleming | 2008-11-29 10:58:29 -0600 (Sat, 29 Nov
2008) | 7 lines update dev-mode compiler flags to match the ones
used by default on Ubuntu Intrepid, so all developers will see
the same warnings and errors since this branch already had some
printf format attributes, enable checking for them and tag
functions that didn't have them format attributes in a consistent
way
------------------------------------------------------------------------
in addition: move some format attributes from main/utils.c to the
header files they belong in, and fix up references to the
relevant functions based on new compiler warnings
* Makefile, funcs/func_sprintf.c (added), main/Makefile,
channels/misdn/ie.c, funcs/func_strings.c, UPGRADE.txt,
res/res_config_sqlite.c, channels/misdn_config.c, funcs/Makefile:
we can now build with -Wformat=2, which found a couple of real
bugs because SPRINTF() use non-literal format strings (which
cannot be checked), move it into its own module so the rest of
func_strings can benefit from format string checking
2008-11-28 14:20 +0000 [r159734] Michiel van Baak <michiel@vanbaak.info>
* res/Makefile: Make res_config_ldap compile with the official
OpenLDAP 2.3.X versions. They removed the LDAP_DEPRECATED define
from their source and since we are using a couple of deprecated
function calls we should define it with a CFLAG. Tested by me on
OpenBSD 4.4 and snuff-home on Linux to make sure everything keeps
compiling. It shouldn't break, we only define the LDAP_DEPRECATED
with this which is what all 2.2.X and older versions of OpenLDAP
did in their own tree.
2008-11-27 20:29 +0000 [r159701] Philippe Sultan <philippe.sultan@gmail.com>
* res/res_jabber.c: Removed duplicate code
2008-11-26 22:11 +0000 [r159664-159666] Russell Bryant <russell@digium.com>
* main/pbx.c: Make a formatting change to test a new post-commit
hook for reviewboard. http://reviewboard.digium.com/r/65/
* main/pbx.c: Make a formatting change to test a new post-commit
hook for reviewboard. http://reviewboard.digium.com/r/65/
* main/pbx.c: Make a formatting change to test a new post-commit
hook for reviewboard. http://reviewboard.digium.com/r/65/
2008-11-26 21:20 +0000 [r159629-159631] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk/agi.h, configure,
include/asterisk/autoconfig.h.in, contrib/asterisk-ng-doxygen,
autoconf/ast_gcc_attribute.m4, configure.ac, res/res_agi.c,
apps/app_stack.c, include/asterisk/optional_api.h (added):
improve handling of API calls provided by loaded modules through
use of some GCC features; this makes app_stack's usage of AGI
APIs even cleaner, and will allow it to work 'as expected' either
with or without res_agi being loaded reviewed at
http://reviewboard.digium.com/r/62
* main/manager.c, CHANGES: add support for event suppression for
AMI-over-HTTP
2008-11-26 19:57 +0000 [r159554] Mark Michelson <mmichelson@digium.com>
* apps/app_dial.c: Add some necessary hangup commands in the case
that forwarding a call fails 1) Hang up the original destination
if the local channel cannot be requested. 2) Hang up the local
channel (in addition to the original destination) if ast_call
fails when calling the newly created local channel. This prevents
channels from sticking around forever in the case of a botched
call forward (e.g. to an extension which does not exist). (closes
issue #13764) Reported by: davidw Patches: 13764_v2.patch
uploaded by putnopvut (license 60) Tested by: putnopvut, davidw
2008-11-26 19:08 +0000 [r159534] Kevin P. Fleming <kpfleming@digium.com>
* agi/Makefile, utils/Makefile, /, Makefile.moddir_rules,
Makefile.rules: Merged revisions 159476 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r159476 | kpfleming | 2008-11-26 12:36:24 -0600 (Wed, 26 Nov
2008) | 7 lines simplify (and slightly bug-fix) the recent
developer-oriented COMPILE_DOUBLE mode ensure that 'make clean'
removes dependency files for .i files that are created in
COMPILE_DOUBLE mode ........
2008-11-26 18:33 +0000 [r159475] Tilghman Lesher <tlesher@digium.com>
* main/udptl.c: If the config file does not exist, then the first
use crashes Asterisk. (closes issue #13848) Reported by:
klaus3000 Patches: udptl.c.patch uploaded by eliel (license 64)
Tested by: blitzrage
2008-11-26 14:58 +0000 [r159437] Mark Michelson <mmichelson@digium.com>
* channels/chan_agent.c: Don't allow for configuration options to
overwrite options set via channel variables on a reload. (closes
issue #13921) Reported by: davidw Patches: 13921.patch uploaded
by putnopvut (license 60) Tested by: davidw
2008-11-26 03:18 +0000 [r159402] Jeff Peeler <jpeeler@digium.com>
* main/features.c: Always parse arguments in park_call_exec so that
app_args is valid. This prevents a crash when executing Park from
the dialplan with no arguments.
2008-11-25 23:03 +0000 [r159360] Steve Murphy <murf@digium.com>
* main/cdr.c, /, channels/chan_iax2.c: Merged revisions 159316 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r159316 | murf | 2008-11-25 15:41:10 -0700 (Tue, 25 Nov 2008) |
15 lines (closes issue #12694) Reported by: yraber Patches:
12694.2nd.diff uploaded by murf (license 17) Tested by: murf,
laurav Thanks to file (Joshua Colp) for his IAX fix. the change
to cdr.c allows no-answer to percolate up into CDR's, and feels
like the right place to locate this fix; if BUSY is done here,
no-answer should be, too. ........
2008-11-25 22:45 +0000 [r159276-159317] Tilghman Lesher <tlesher@digium.com>
* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample,
include/asterisk/dsp.h, CHANGES, main/dsp.c: Add an option,
waitfordialtone, for UK analog lines which do not end a call
until the originating line hangs up. (closes issue #12382)
Reported by: one47 Patches:
zap-waitfordialtone-trunk.080901.patch uploaded by one47 (license
23) zap-waitfordialtone-bra-1.4.21.2.patch uploaded by fleed
(license 463) Tested by: fleed
* /, channels/chan_iax2.c: Merged revisions 159269 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r159269 | tilghman | 2008-11-25 15:56:48 -0600 (Tue, 25
Nov 2008) | 7 lines Don't try to send a response on a NULL pvt.
(closes issue #13919) Reported by: barthpbx Patches:
chan_iax2.c.patch uploaded by eliel (license 64) Tested by:
barthpbx ........
2008-11-25 21:49 +0000 [r159250] Mark Michelson <mmichelson@digium.com>
* apps/app_followme.c: Make the options for the general and
profiles more consistent for the "pls_hold_prompt" option. This
does not affect any released version of Asterisk, so there is no
need to update the CHANGES file for this. (closes issue #13893)
Reported by: eliel
2008-11-25 21:42 +0000 [r159162-159247] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_iax2.c: Merged revisions 159246 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
................ r159246 | tilghman | 2008-11-25 15:40:28 -0600
(Tue, 25 Nov 2008) | 14 lines Merged revisions 159245 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r159245 | tilghman | 2008-11-25 15:37:06 -0600 (Tue, 25 Nov 2008)
| 7 lines Regression fix for last security fix. Set the iseqno
correctly. (closes issue #13918) Reported by: ffloimair Patches:
20081119__bug13918.diff.txt uploaded by Corydon76 (license 14)
Tested by: ffloimair ........ ................
* pbx/pbx_realtime.c: Don't actually do anything with a negative
priority, because we ignore it in the result, anyway.
* main/pbx.c: Don't limit the length of the hint at the final step
(from ~8100 chars max (or ~500 chars max on LOW_MEMORY) to 80
chars max). This will allow more channels to be used in a single
hint.
2008-11-25 16:18 +0000 [r159093] Terry Wilson <twilson@digium.com>
* apps/app_festival.c: Add missing variable declaration for PPC
code
2008-11-25 05:19 +0000 [r159050-159054] Tilghman Lesher <tlesher@digium.com>
* apps/app_readexten.c: Copyright clarification; also, have
variable set to "t" or "i" on timeout or invalid extension,
respectively. (closes issue #13944) Reported by: chappell
* channels/chan_usbradio.c, /, configure,
include/asterisk/autoconfig.h.in, configure.ac,
channels/xpmr/xpmr.c, apps/app_rpt.c: Merged revisions 159025 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r159025 | tilghman | 2008-11-24 22:50:00 -0600 (Mon, 24 Nov 2008)
| 3 lines System call ioperm is non-portable, so check for its
existence in autoconf. (Closes issue #13863) ........
2008-11-25 03:49 +0000 [r158992] Terry Wilson <twilson@digium.com>
* channels/chan_usbradio.c: Make chan_usbradio compile under dev
mode
2008-11-25 01:01 +0000 [r158959] Sean Bright <sean.bright@gmail.com>
* funcs/func_dialgroup.c, channels/chan_sip.c,
include/asterisk/astobj2.h, res/res_phoneprov.c,
main/taskprocessor.c, channels/chan_console.c,
channels/chan_iax2.c, apps/app_queue.c, main/astobj2.c,
main/config.c, main/manager.c, res/res_timing_pthread.c,
main/features.c, res/res_timing_timerfd.c, utils/hashtest2.c,
res/res_clialiases.c: This is basically a complete rollback of
r155401, as it was determined that it would be best to maintain
API compatibility. Instead, this commit introduces
ao2_callback_data() which is functionally identical to
ao2_callback() except that it allows you to pass arbitrary data
to the callback. Reviewed by Mark Michelson via ReviewBoard:
http://reviewboard.digium.com/r/64
2008-11-25 00:19 +0000 [r158876-158925] Matthew Nicholson <mnicholson@digium.com>
* main/file.c: Fix compiling in dev mode.
* UPGRADE.txt, apps/app_queue.c: Make the Join event from app_queue
use CallerIDNum insead of CallerID for indicating the callerid
number just like the rest of asterisk. (closes issue #13883)
Reported by: davidw
* main/manager.c, res/res_agi.c, include/asterisk/manager.h: Added
EVENT_FLAG_AGI and used it for manager calls in res_agi.c (closes
issue #13873) Reported by: fnordian Patches: ami_agievent.patch
uploaded by fnordian (license 110)
2008-11-24 21:52 +0000 [r158857] Tilghman Lesher <tlesher@digium.com>
* main/dsp.c: Add a bit of documentation (thanks, I-MOD) on what
the silence threshold constant actually does and what values are
valid for it.
2008-11-24 21:27 +0000 [r158851] Matthew Nicholson <mnicholson@digium.com>
* main/file.c: Make ast_streamfile() check the result of
ast_openstream() before doing anything with it. (closes issue
#13955) Reported by: chris-mac
2008-11-24 18:11 +0000 [r158808] Terry Wilson <twilson@digium.com>
* apps/app_minivm.c: This patch adds a new application for sending
MWI to phones via Asterisk's event subsystem. Also, the minivm
documentation is all converted to use xmldocs. (closes issue
#13946) Reported by: Marquis Patches:
minivmmwi_plus_xmldocs.patch uploaded by Marquis (license 32)
Tested by: otherwiseguy, Marquis
2008-11-23 03:36 +0000 [r158754-158756] Sean Bright <sean.bright@gmail.com>
* channels/chan_sip.c, configs/sip.conf.sample: If you enabled
'notifycid' one of the limitations is that the calling channel is
only found if it dialed the extension that was subscribed to. You
can now specify 'ignore-context' for the 'notifycid' option in
sip.conf which will, as it's value implies, ignore the current
context of the caller when doing the lookup.
* channels/chan_sip.c: No need to use a separate structure for this
since we can just pass our sip_pvt pointer in directly.
2008-11-22 17:17 +0000 [r158686-158723] Michiel van Baak <michiel@vanbaak.info>
* funcs/func_realtime.c: last commit worked on OpenBSD but still
generated warning on Ubuntu. Initialise a variable so
--enable-dev-mode does not complain
* channels/chan_skinny.c: dont send reorder tone after a device is
hungup if a dialout is abandoned or failed. Without this reorder
tone will play after hangup and both wedhorn's and my wife have
threatened to use an axe on our asterisk box (closes issue
#13948) Reported by: wedhorn Patches: switch.diff uploaded by
wedhorn (license 30)
* channels/chan_skinny.c: Add Media Source Update to skinny's
control2str (issue #13948)
* channels/chan_skinny.c: fix a very occasional core dump in
chan_skinny found by wedhorn. (issue #13948)
* funcs/func_realtime.c: make this compile under devmode
2008-11-21 23:40 +0000 [r158606] Steve Murphy <murf@digium.com>
* /, main/features.c: Merged revisions 158603 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r158603 | murf | 2008-11-21 16:14:50 -0700 (Fri, 21 Nov 2008) |
11 lines In reference to the fix made for 13871, I was merging
the fix into 1.6.0 and realized I missed the code in the h-exten
block, and didn't catch it because my test case had the h-exten
commented out. So, this corrects the code I missed, as a
preventative against another crash report. Tested with the
h-exten defined, all is well. ........
2008-11-21 23:33 +0000 [r158602-158605] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c: Allow space within an extension, when the space is
within a character class. (requested by lmadsen on -dev, patch by
me)
* main/pbx.c, /: Merged revisions 158600 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r158600 | tilghman | 2008-11-21 17:07:46 -0600 (Fri, 21 Nov 2008)
| 5 lines The passed extension may not be the same in the list as
the current entry, because we strip spaces when copying the
extension into the structure. Therefore, use the copied item to
place the item into the list. (found by lmadsen on -dev, fixed by
me) ........
2008-11-21 22:12 +0000 [r158540] Russell Bryant <russell@digium.com>
* /, include/asterisk/astobj2.h, main/astobj2.c: Merged revisions
158539 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r158539 | russell | 2008-11-21 16:05:55 -0600 (Fri, 21 Nov 2008)
| 2 lines When compiling with DEBUG_THREADS, report the real
file/func/line for ao2_lock/ao2_unlock ........
2008-11-21 21:47 +0000 [r158484] Steve Murphy <murf@digium.com>
* /, main/features.c: Merged revisions 158483 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r158483 | murf | 2008-11-21 14:19:47 -0700 (Fri, 21 Nov 2008) |
11 lines (closes issue #13871) Reported by: mdu113 This one is
totally my fault. The code doesn't even create a bridge CDR if
the channel CDR has POST_DISABLED. I didn't check for that at the
end of the bridge. Fixed with a few small insertions. Tested.
Looks good. No cdr generated, no crash, no unnecc. data objects
created either. ........
2008-11-21 21:06 +0000 [r158482] Matthew Fredrickson <creslin@digium.com>
* channels/chan_dahdi.c: Fix for #13963. Make physical channel
mapping unconfigured default
2008-11-21 20:42 +0000 [r158449] Kevin P. Fleming <kpfleming@digium.com>
* UPGRADE-1.2.txt, UPGRADE-1.4.txt, UPGRADE.txt, UPGRADE-1.6.txt,
CHANGES: as suggested by jtodd, document the purposes of the
CHANGES and UPGRADE files
2008-11-21 19:40 +0000 [r158414] Jason Parker <jparker@digium.com>
* main/manager.c: Make sure we add the Event header for
CoreShowChannels. (closes issue #13334) Reported by: srt Patches:
13334_missing_event_header_in_core_show_channel.diff uploaded by
srt (license 378)
2008-11-21 17:08 +0000 [r158374] Terry Wilson <twilson@digium.com>
* cdr/cdr_csv.c: Reloading the config and having no changes still
initialized some settings to 0. Initialize settings after doing
all of the cfg checks. (closes issue #13942) Reported by: davidw
Patches: cdr_diff.txt uploaded by otherwiseguy (license 396)
Tested by: davidw
2008-11-21 15:53 +0000 [r158315] Doug Bailey <dbailey@digium.com>
* channels/chan_sip.c: Add fix to prevent crash during reload if
there is an outstanding MWI registration message pending.
2008-11-21 01:22 +0000 [r158230-158266] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Use a more expressive constant for a 64-bit
scanned int
* channels/chan_sip.c: Use some magic constants to get the right
size for this sscanf statement. Thanks Richard!
* channels/chan_sip.c: Fix the build for 32-bit systems. %lu is
only 32-bits on 32-bit systems, so we need to use %llu instead.
Of course %llu is 128-bits on 64-bit systems, so we have to cast
to unsigned long long. No harm, but it's sure annoying.
* channels/chan_sip.c: Change the remote user agent session version
variable from an int to a uint64_t. This prevents potential
comparison problems from happening if the version string exceeds
INT_MAX. This was an apparent problem for one user who could not
properly place a call on hold since the version in the SDP of the
re-INVITE to place the call on hold greatly exceeded INT_MAX.
This also aligns with RFC 2327 better since it recommends using
an NTP timestamp for the version (which is a 64-bit number).
(closes issue #13531) Reported by: sgofferj Patches: 13531.patch
uploaded by putnopvut (license 60) Tested by: sgofferj
2008-11-20 19:41 +0000 [r158188] Sean Bright <sean.bright@gmail.com>
* res/ael/pval.c: Fix one case where the application argument was
not converted from a pipe to a comma. This was causing problems
with switch statements with empty expressions. (closes issue
#13901) Reported by: smurfix Patches: 20081118_bug13901.diff
uploaded by seanbright (license 71) Tested by: seanbright
Reviewed by: murf
2008-11-20 18:20 +0000 [r158082-158133] Mark Michelson <mmichelson@digium.com>
* include/asterisk/file.h, main/frame.c, /, channels/chan_sip.c,
main/file.c, include/asterisk/frame.h: Merged revisions 158072
via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r158072 | twilson | 2008-11-20 11:48:58 -0600 (Thu, 20
Nov 2008) | 2 lines Begin on a crusade to end trailing
whitespace! ........
* /, channels/chan_sip.c: Merged revisions 158071 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r158071 | mmichelson | 2008-11-20 11:48:42 -0600 (Thu, 20 Nov
2008) | 16 lines We don't handle 4XX responses to BYE well.
According to section 15 of RFC 3261, we should terminate a dialog
if we receive a 481 or 408 in response to our BYE. Since I am
aware of at least one phone manufacturer who may sometimes send a
404 as well, I am being liberal and saying that any 4XX response
to a BYE should result in a terminated dialog. (closes issue
#12994) Reported by: pabelanger Patches: 12994.patch uploaded by
putnopvut (license 60) Closes AST-129 ........
2008-11-20 17:53 +0000 [r158078] Ryan Brindley <rbrindley@digium.com>
* main/config.c: more formatting corrections :: one line for loops
and if statements still need {}
2008-11-20 17:48 +0000 [r158072] Terry Wilson <twilson@digium.com>
* cdr/cdr_sqlite3_custom.c, cdr/cdr_sqlite.c, cdr/Makefile,
cdr/cdr_adaptive_odbc.c, cdr/cdr_pgsql.c, cdr/cdr_odbc.c,
cdr/cdr_radius.c, cdr/cdr_custom.c, cdr/cdr_manager.c,
cdr/cdr_csv.c: Begin on a crusade to end trailing whitespace!
2008-11-20 17:46 +0000 [r158070] Ryan Brindley <rbrindley@digium.com>
* main/config.c: formatting changes :: one line for loops and if
statements should have {}
2008-11-20 17:39 +0000 [r158066] Mark Michelson <mmichelson@digium.com>
* apps/app_dial.c, /, channels/chan_sip.c: Merged revisions 158053
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r158053 | mmichelson | 2008-11-20 11:33:06 -0600 (Thu, 20 Nov
2008) | 12 lines Make sure to set the hangup cause on the calling
channel in the case that ast_call() fails. For incoming SIP
channels, this was causing us to send a 603 instead of a 486 when
the call-limit was reached on the destination channel. (closes
issue #13867) Reported by: still_nsk Patches: 13867.diff uploaded
by putnopvut (license 60) Tested by: blitzrage ........
2008-11-20 17:37 +0000 [r158062] Jeff Peeler <jpeeler@digium.com>
* main/file.c: (closes issue #12929) Reported by: snyfer This
handles the case for a zero length file to attempt to be
streamed. Instead of failing from not playing any data, go ahead
and return success as ast_streamfile should consider playing
nothing a success when there is nothing to play.
2008-11-20 17:37 +0000 [r158061] Jason Parker <jparker@digium.com>
* README: Whitespace fix
2008-11-20 00:08 +0000 [r157974] Kevin P. Fleming <kpfleming@digium.com>
* main/stdtime, /, main/db1-ast/hash, codecs/gsm/Makefile,
Makefile.moddir_rules, main/db1-ast/db, channels/misdn,
main/db1-ast/mpool, res/ais, res/Makefile, pbx/Makefile,
Makefile.rules, res/snmp, main/stdtime/Makefile, codecs/gsm/src,
main/db1-ast/btree, channels/misdn/Makefile, main/db1-ast/recno,
res/ael, pbx/ael, channels, main/db1-ast/Makefile: Merged
revisions 157859 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r157859 | kpfleming | 2008-11-19 15:34:47 -0600 (Wed, 19 Nov
2008) | 7 lines the gcc optimizer frequently finds broken code
(use of uninitalized variables, unreachable code, etc.), which is
good. however, developers usually compile with the optimizer
turned off, because if they need to debug the resulting code,
optimized code makes that process very difficult. this means that
we get code changes committed that weren't adequately checked
over for these sorts of problems. with this build system change,
if (and only if) --enable-dev-mode was used and DONT_OPTIMIZE is
turned on, when a source file is compiled it will actually be
preprocessed (into a .i or .ii file), then compiled once with
optimization (with the result sent to /dev/null) and again
without optimization (but only if the first compile succeeded, of
course). while making these changes, i did some cleanup work in
Makefile.rules to move commonly-used combinations of flag
variables into their own variables, to make the file easier to
read and maintain ........
2008-11-20 00:06 +0000 [r157973] Terry Wilson <twilson@digium.com>
* res/res_timing_timerfd.c: Fix compiling
2008-11-19 23:30 +0000 [r157906-157940] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Add a space to the output
* apps/app_queue.c: Add a RES_NOT_DYNAMIC case for the CLI command
'queue remove member'
* CHANGES: Commit CHANGES change I promised when submitting
res_timing_timerfd
2008-11-19 22:01 +0000 [r157893] Tilghman Lesher <tlesher@digium.com>
* CHANGES: Add info about REALTIME_FIELD and REALTIME_HASH
2008-11-19 21:55 +0000 [r157874] Mark Michelson <mmichelson@digium.com>
* res/res_timing_timerfd.c: Cast this value since a uint64_t is not
the same as an unsigned long long on a 64-bit machine. Reported
by kpfleming on IRC
2008-11-19 21:54 +0000 [r157870] Tilghman Lesher <tlesher@digium.com>
* funcs/func_realtime.c: Two new functions, REALTIME_FIELD, and
REALTIME_HASH, which should make querying realtime from the
dialplan a little more consistent and easy to use. The original
REALTIME function is preserved, for those who are already
accustomed to that interface. (closes issue #13651) Reported by:
Corydon76 Patches: 20081119__bug13651__2.diff.txt uploaded by
Corydon76 (license 14) Tested by: blitzrage, Corydon76
2008-11-19 19:37 +0000 [r157820] Mark Michelson <mmichelson@digium.com>
* build_tools/menuselect-deps.in, configure,
include/asterisk/autoconfig.h.in, res/res_timing_pthread.c,
configure.ac, res/res_timing_dahdi.c, res/res_timing_timerfd.c
(added), makeopts.in: Merge the changes from the
res_timing_timerfd branch. This provides a new timing interface.
In order to use it, you must be running a Linux with a kernel
version of 2.6.25 or newer and glibc 2.8 or newer. This timing
interface is a good alternative if a timing source is necessary
(e.g. for IAX trunking) but DAHDI is otherwise unnecessary for
the system. For now, this commit contains the actual work done in
the res_timing_timerfd branch. There are no notices in the README
or CHANGES files yet, but they will be added in my next commit.
The timing API of Asterisk also needs to have a bit of work done
with regards to choosing which timing interface to use. This
commit makes the choice a build-time decision, by only allowing
one of the timer interfaces to be chosen in menuselect. It would
be preferable if the choice could be made at run-time, however.
The preferred timing interface could be loaded and tested, and if
it does not work, choice number two may be used instead. That
sort of thing. That is beyond the scope of work in this branch
though.
2008-11-19 19:25 +0000 [r157818] Terry Wilson <twilson@digium.com>
* channels/chan_vpb.cc, cdr/cdr_sqlite3_custom.c,
channels/iax2-provision.c, cdr/cdr_adaptive_odbc.c,
cdr/cdr_pgsql.c, cdr/cdr_radius.c, cdr/cdr_tds.c,
channels/misdn_config.c, cdr/cdr_csv.c, channels/chan_usbradio.c,
channels/chan_skinny.c, main/logger.c, res/ais/evt.c,
pbx/pbx_dundi.c, cdr/cdr_odbc.c, cdr/cdr_custom.c,
cdr/cdr_manager.c, main/xmldoc.c, res/res_clialiases.c: Fix
checking for CONFIG_STATUS_FILEINVALID so that modules don't
crash upon trying to parse an invalid config
2008-11-19 18:28 +0000 [r157784] Tilghman Lesher <tlesher@digium.com>
* configure, configure.ac: Add check for t38_terminal_init in
spandsp (not found in 0.0.6, so it should fail reasonably)
(closes issue #13473) Reported by: genie Patches:
20080916__bug13473.diff.txt uploaded by Corydon76 (license 14)
2008-11-19 13:45 +0000 [r157706-157743] Kevin P. Fleming <kpfleming@digium.com>
* res/res_agi.c: correct small bug introduced during API conversion
* UPGRADE.txt, UPGRADE-1.6.txt: move relevant entries into
UPGRADE.txt and resync UPGRADE-1.6.txt with previous branches
* include/asterisk/agi.h, res/res_agi.c, UPGRADE.txt,
UPGRADE-1.6.txt (added), apps/app_stack.c: make some corrections
to the ast_agi_register_multiple(), ast_agi_unregister_multiple()
and ast_agi_fdprintf() API calls to be consistent with API
guidelines also, move UPGRADE.txt to UPGRADE-1.6.txt and make the
new UPGRADE.txt contain information about upgrading between
Asterisk 1.6 releases
2008-11-19 05:37 +0000 [r157675] Terry Wilson <twilson@digium.com>
* configs/cdr_adaptive_odbc.conf.sample: Comment out config line
that is in a commented out context
2008-11-19 01:02 +0000 [r157639] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/logger.h, main/logger.c, main/utils.c,
include/asterisk/strings.h: Starting with a change to ensure that
ast_verbose() preserves ABI compatibility in 1.6.1 (as compared
to 1.6.0 and versions of 1.4), this change also deprecates the
use of Asterisk with FreeBSD 4, given the central use of va_copy
in core functions. va_copy() is C99, anyway, and we already
require C99 for other purposes, so this isn't really a big change
anyway. This change also simplifies some of the core ast_str_*
functions.
2008-11-19 00:59 +0000 [r157632] Mark Michelson <mmichelson@digium.com>
* main/astmm.c: If malloc returns NULL, we need to return NULL
immediately or else Asterisk will crash when attempting to
dereference the NULL pointer (closes issue #13858) Reported by:
eliel Patches: astmm.c.patch uploaded by eliel (license 64)
2008-11-19 00:27 +0000 [r157600] Sean Bright <sean.bright@gmail.com>
* Makefile, build_tools/make_version, configure, configure.ac,
build_tools/make_buildopts_h, makeopts.in: Fix a few build
problems on Solaris (and check for an md5 utility in configure
instead of the icky loop I was doing before). (closes issue
#13842) Reported by: snuffy Patches: bug13842_20081106.diff
uploaded by snuffy (license 35) 13842.diff uploaded by seanbright
(license 71) Tested by: snuffy
2008-11-18 23:59 +0000 [r157496-157592] Mark Michelson <mmichelson@digium.com>
* res/res_musiconhold.c: This change prevents a crash from
occurring if res_musiconhold.so is unloaded and then Asterisk is
stopped. The problem was that we are not unregistering the
ast_moh_destroy function at exit. (closes issue #13761) Reported
by: eliel Patches: res_musiconhold.c.patch uploaded by eliel
(license 64)
* Makefile: Add some missing $(DESTDIR)s to the bininstall target
of the Makefile. (closes issue #13875) Reported by: pabelanger
Patches: Makefile.155928 uploaded by pabelanger (license 224)
* apps/app_voicemail.c: Fix the logic for when delete=yes when IMAP
storage is in use so that the message is deleted from both local
and IMAP storage. (closes issue #13642) Reported by: jaroth
Patches: deleteyes.patch uploaded by jaroth (license 50)
* channels/chan_sip.c: Merged revisions 157503 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r157503 | mmichelson | 2008-11-18 16:47:57 -0600 (Tue, 18 Nov
2008) | 13 lines Add some missing invite state changes necessary
in the sip_write function. Not setting the invite state correctly
on the call was resulting in the Record application leaving empty
files. I also have updated the doxygen comment next to the
declaration of the INV_EARLY_MEDIA constant to reflect that we
also use this state when we *send* a 18X response to an INVITE.
(closes issue #13878) Reported by: nahuelgreco Patches:
sip-early-media-recording-1.4.22.patch uploaded by nahuelgreco
(license 162) Tested by: putnopvut ........
* channels/chan_sip.c: Based on Russell's advice on the
asterisk-dev list, I have changed from using a global lock in
update_call_counter to using the locks within the sip_pvt and
sip_peer structures instead.
2008-11-18 21:15 +0000 [r157460-157463] Jason Parker <jparker@digium.com>
* Makefile: Remove echo line that is unnecessary (Thanks
seanbright).
* contrib/init.d/rc.archlinux.asterisk: Make this executable
* Makefile, contrib/init.d/rc.archlinux.asterisk (added): Add init
script for ArchLinux (closes issue #13667) Reported by: sherif
Patches: archlinux_rc_makefile.patch uploaded by sherif (license
591) archlinux_rc_makefile-2.patch uploaded by mvanbaak (license
7)
2008-11-18 20:23 +0000 [r157427] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: * Add a lock to be used in the
update_call_counter function. * Revert logic to mirror 1.4's in
the sense that it will not allow the call counter to dip below 0.
These two measures prevent potential races that could cause a SIP
peer to appear to be busy forever. (closes issue #13668) Reported
by: mjc Patches: hintfix_trunk_rev152649.patch uploaded by
wolfelectronic (license 586)
2008-11-18 19:16 +0000 [r157366] Jeff Peeler <jpeeler@digium.com>
* /, apps/app_meetme.c: Merged revisions 157365 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r157365 | jpeeler | 2008-11-18 13:13:33 -0600 (Tue, 18 Nov 2008)
| 6 lines (closes issue #13899) Reported by: akkornel This fix is
the result of a bug fix in ast_app_separate_args r124395. If an
argument does not exist it should always be set to a null string
rather than a null pointer. ........
2008-11-18 18:31 +0000 [r157306] Mark Michelson <mmichelson@digium.com>
* apps/app_dial.c, channels/chan_local.c, /, main/features.c,
include/asterisk/channel.h, apps/app_followme.c: Merged revisions
157305 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r157305 | mmichelson | 2008-11-18 12:25:55 -0600 (Tue, 18 Nov
2008) | 12 lines Fix a crash in the end_bridge_callback of
app_dial and app_followme which would occur at the end of an
attended transfer. The error occurred because we initially stored
a pointer to an ast_channel which then was hung up due to a
masquerade. This commit adds a "fixup" callback to the
bridge_config structure to allow for end_bridge_callback_data to
be changed in the case that a new channel pointer is needed for
the end_bridge_callback. ........
2008-11-18 18:07 +0000 [r157302] Steve Murphy <murf@digium.com>
* main/config.c: (closes issue #13420) Reported by: alex70 Patches:
13420.13539.patch uploaded by murf (license 17) Tested by: murf,
awk This fixes two problems: a spurious linefeed insertion
probably left over from pre-precomment times. Only generated when
category had no previous comments. The other problem: Insertions
could get the line-numbering out of whack and generate negative
line numbers, causing chunks of line numbers to be emitted, on
the scale of the number of lines up to that point in the file. In
such cases, abort the looping, and all is well.
2008-11-17 22:25 +0000 [r157253] Tilghman Lesher <tlesher@digium.com>
* apps/app_dial.c: Can't use items duplicated off the stack frame
in an element returned from a function: in these cases, we have
to use the heap, or garbage will result. (closes issue #13898)
Reported by: alecdavis Patches: 20081114__bug13898__2.diff.txt
uploaded by Corydon76 (license 14) Tested by: alecdavis
2008-11-15 19:51 +0000 [r157105-157167] Kevin P. Fleming <kpfleming@digium.com>
* Makefile.rules: ensure that if a .i file (preprocessed source) is
present, the .o file is made from it, not from the .c file (this
only works because GNU makes respects the order the rules are
defined)
* Makefile, /, Makefile.moddir_rules, Makefile.rules: Merged
revisions 157162-157163 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r157162 | kpfleming | 2008-11-15 20:24:24 +0100 (Sat, 15 Nov
2008) | 1 line dist-clean should remove dependency information
files as well ........ r157163 | kpfleming | 2008-11-15 20:31:03
+0100 (Sat, 15 Nov 2008) | 1 line when an individual directory
dist-clean is run, run clean in that directory first, and when
running top-level dist-clean, do not run subdirectory clean
operations twice ........
* /, contrib/asterisk-ng-doxygen: Merged revisions 157104 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r157104 | kpfleming | 2008-11-15 19:00:32 +0100 (Sat, 15 Nov
2008) | 13 lines major update to doxygen configuration file: 1)
update to doxygen 1.5.x style file, as used in trunk 2) tell
doxygen where are header files are, so include-file processing
can be done 3) make all macros that are used to define
variables/functions be expanded, so that doxygen will properly
document the resulting variable/function 4) make all macros that
are used to provide the contents of a variable (structure) be
expanded, so that doxygen will be able to document the resulting
fields 5) suppress compiler attributes (__attribute__(xxx)) from
being seen by doxygen, so it will properly match up function
definition and usage (for an example of th effect of this, look
at the doxygen docs for ast_log() from before and afte this
commit) ........
2008-11-15 15:37 +0000 [r157073] Eliel C. Sardanons <eliels@gmail.com>
* main/xmldoc.c: Avoid a not needed cast, making code more
readable.
2008-11-15 04:25 +0000 [r157039-157041] Russell Bryant <russell@digium.com>
* channels/chan_sip.c, main/features.c, main/taskprocessor.c: Fix a
few more places where the case insensitive hash should be used
since the comparison is case insensitive.
* channels/chan_console.c: Use the new case insensitive hash
function for console interfaces. The comparison function is case
insensitive.
2008-11-14 22:36 +0000 [r157006] Tilghman Lesher <tlesher@digium.com>
* cdr/cdr_adaptive_odbc.c, configs/cdr_adaptive_odbc.conf.sample:
Allow setting static values in CDRs
2008-11-14 21:19 +0000 [r156962] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Revision 155513 of chan_sip.c in trunk
inadvertently removed a very important line to set the "len"
field for incoming SIP requests. The result was that all incoming
SIP messages appeared to be 0-length, meaning Asterisk could do
no meaningful processing of anything SIP-related
2008-11-14 17:35 +0000 [r156916-156918] Terry Wilson <twilson@digium.com>
* res/res_phoneprov.c: Cleanup whitespace issues
* res/res_phoneprov.c: Use Mark's new ast_str_case_hash function
instead of jumping through hoops to do insensitive case lookups
2008-11-14 17:02 +0000 [r156911] Tilghman Lesher <tlesher@digium.com>
* main/manager.c: Ping is missing the standard double-newline after
the event. (closes issue #13903) Reported by: kebl0155
2008-11-14 16:53 +0000 [r156883] Mark Michelson <mmichelson@digium.com>
* UPGRADE.txt, include/asterisk/strings.h, apps/app_queue.c: Fix
some refcounting in app_queue.c and change the hashing used by
app_queue.c to be case-insensitive. This is accomplished by
adding a new case-insensitive hashing function. This was
necessary to prevent bad refcount errors (and potential crashes)
which would occur due to the fact that queues were initially read
from the config file in a case-sensitive manner. Then, when a
user issued a CLI command or manager action, we allowed for
case-insensitive input and used that input to directly try to
find the queue in the hash table. The result was either that we
could not find a queue that was input or worse, we would end up
hashing to a completely bogus value based on the input. This
commit resolves the problem presented in issue #13703. However,
that issue was reported against 1.6.0. Since this fix introduces
a behavior change, I am electing to not place this same fix in to
the 1.6.0 or 1.6.1 branches, and instead will opt for a change
which does not change behavior.
2008-11-14 16:34 +0000 [r156874] Matthew Fredrickson <creslin@digium.com>
* channels/chan_dahdi.c: Remove some useless locking and make sure
we hangup channels on a link when we get a GRS.
2008-11-14 15:20 +0000 [r156817] Mark Michelson <mmichelson@digium.com>
* /, apps/app_voicemail.c: Merged revisions 156816 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r156816 | mmichelson | 2008-11-14 09:18:59 -0600 (Fri,
14 Nov 2008) | 10 lines If the prompt to reenter a voicemail
password timed out, it resulted in the password not being saved,
even if the input matched what you gave when first prompted to
enter a new password. This is because the return value of
ast_readstring was checked, but not checked properly. This bug
was discovered by Jared Smith during an Asterisk training course.
Thanks for reporting it! ........
2008-11-14 00:43 +0000 [r156690-156756] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_while.c: Merged revisions 156755 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r156755 | tilghman | 2008-11-13 18:41:37 -0600 (Thu, 13 Nov 2008)
| 6 lines ast_waitfordigit() requires that the channel be up, for
no good logical reason. This prevents While/EndWhile from working
within the "h" extension. Reported by: jgalarneau (for ABE C.2)
Fixed by: me ........
* main/manager.c, /: Merged revisions 156688 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r156688 | tilghman | 2008-11-13 15:24:00 -0600 (Thu, 13 Nov 2008)
| 7 lines Provide more space for all the data which can appear in
an originating channel name. (closes issue #13398) Reported by:
bamby Patches: manager.c.diff uploaded by bamby (license 430)
........
2008-11-13 19:17 +0000 [r156649] Jeff Peeler <jpeeler@digium.com>
* main/pbx.c: (closes issue #13891) Reported by: smurfix This
reverts a change I made in 116297. At the time it seemed the
change was required to solve an issue with attempting a transfer
but then letting it timeout without dialing any digits. However,
I didn't realize that having an empty extension was possible. I'm
removing the immediate return that was added in
pbx_find_extension if the extension is null.
2008-11-13 19:10 +0000 [r156647] Tilghman Lesher <tlesher@digium.com>
* channels/chan_dahdi.c: Command offsets were not changed correctly
when the command syntax for 'pri set debug' was changed from 'pri
debug'.
2008-11-13 17:07 +0000 [r156612] Mark Michelson <mmichelson@digium.com>
* configure, autoconf/ast_c_compile_check.m4: Kevin sent a note
indicating that this change is not necessary, so I am reverting
it
2008-11-13 15:46 +0000 [r156535-156575] Eliel C. Sardanons <eliels@gmail.com>
* apps/app_meetme.c, doc/appdocsxml.dtd, main/xmldoc.c: Introduce
XML documentation for: - MeetMe() - MeetMeCount() -
MeetMeChannelAdmin() - MeetMeAdmin() - SLAStation() - SLATrunk()
- Add an attribute to optionlist 'hasparams' with the same
functionality as the one we have in <parameter> and <argument>
(the DTD was updated) - Fix a leak when getting an attribute
while parsing an <optionlist>.
* main/xmldoc.c: Fix a typo introduced when changing
xmldoc_has_arguments() to xmldoc_has_inside() we need to pass the
name of the node that we are looking for.
* include/asterisk/xml.h, include/asterisk/xmldoc.h, main/xmldoc.c:
Remove trailing whitespaces using ':%s/\s\+$//' pointed by
seanbright on #asterisk-dev
2008-11-12 23:13 +0000 [r156443] Sean Bright <sean.bright@gmail.com>
* /: Use the reviewboard:url SVN property so post-review will work
without modification.
2008-11-12 21:34 +0000 [r156388] Tilghman Lesher <tlesher@digium.com>
* apps/app_dial.c, /: Merged revisions 156386 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r156386 | tilghman | 2008-11-12 15:18:57 -0600 (Wed, 12 Nov 2008)
| 5 lines When using call limits under 1 second, infinite call
lengths are allowed, instead. (closes issue #13851) Reported by:
ruddy ........
2008-11-12 20:27 +0000 [r156355] Eliel C. Sardanons <eliels@gmail.com>
* res/res_clialiases.c: - Make alias->real_cmd point to the
allocated space outside alias->alias. - Register the aliased cli
command (or we will not alias anything). - Use ARRAY_LEN() when
possible.
2008-11-12 19:47 +0000 [r156299] Steve Murphy <murf@digium.com>
* main/pbx.c, /: Merged revisions 156297 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r156297 | murf | 2008-11-12 12:36:16 -0700 (Wed, 12 Nov 2008) |
18 lines It turns out that the 0x0XX00 codes being returned for
N, X, and Z are off by one, as per conversation with jsmith on
#asterisk-dev; he was teaching a class and disconcerted that this
published rule was not being followed, with patterns _NXX,
_[1-8]22 and _[2-9]22... and NXX was winning, but [1-8] should
have been. This change, tested on these 3 patterns now picks the
proper one. However, this change may surprise users who set up
dialplans based on previous behavior, which has been there for
what, 2 and half years or so now. ........
2008-11-12 19:38 +0000 [r156298] Russell Bryant <russell@digium.com>
* res/res_clialiases.c: Fix a bug caused by using sizeof(pointer)
instead of sizeof(the struct)
2008-11-12 19:28 +0000 [r156295] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_meetme.c: Merged revisions 156294 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r156294 | tilghman | 2008-11-12 13:26:45 -0600 (Wed, 12 Nov 2008)
| 6 lines If the SLA thread is not started, then reload causes a
memory leak. (closes issue #13889) Reported by: eliel Patches:
app_meetme.c.patch uploaded by eliel (license 64) ........
2008-11-12 19:11 +0000 [r156290] Jeff Peeler <jpeeler@digium.com>
* /, apps/app_meetme.c: Merged revisions 156289 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r156289 | jpeeler | 2008-11-12 13:10:12 -0600 (Wed, 12 Nov 2008)
| 3 lines For whatever reason, gcc only warned me about the
possible use of an uninitialized variable when compiling 1.6.1.
........
2008-11-12 18:55 +0000 [r156243] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_iax2.c: Merged revisions 156229 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r156229 | tilghman | 2008-11-12 12:39:21 -0600 (Wed, 12
Nov 2008) | 11 lines Revert revision 132506, since it
occasionally caused IAX2 HANGUP packets not to be sent, and
instead, schedule a task to destroy the iax2 pvt structure 10
seconds later. This allows the IAX2 HANGUP packet to be queued,
transmitted, and ACKed before the pvt is destroyed. (closes issue
#13645) Reported by: dzajro Patches:
20081111__bug13645__3.diff.txt uploaded by Corydon76 (license 14)
Tested by: vazir Reviewed: http://reviewboard.digium.com/r/51/
........
2008-11-12 18:32 +0000 [r156228] Jeff Peeler <jpeeler@digium.com>
* /, apps/app_meetme.c: Merged revisions 156178 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r156178 | jpeeler | 2008-11-12 11:53:44 -0600 (Wed, 12 Nov 2008)
| 8 lines (closes issue #13173) Reported by: pep This change adds
an announce_thread responsible for playing announcements to an
existing conference. This allows all announcing to be immediately
stopped if necessary but more importantly allows other threads
that need to play something to not block. There are multiple
benefits to this, but the actual bug is for solving the scenario
for a channel to be unusable after hang up for the entire
duration of the parting announcement. The parting announcement
can be extremely long depending on what the user recorded upon
joining the conference. Reviewed by Russell on Review Board:
http://reviewboard.digium.com/r/25/ ........
2008-11-12 17:41 +0000 [r156169] Mark Michelson <mmichelson@digium.com>
* apps/app_dial.c, /: Merged revisions 156167 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r156167 | mmichelson | 2008-11-12 11:38:33 -0600 (Wed, 12 Nov
2008) | 7 lines When doing some tests, I was having a crash at
the end of every call if an attended transfer occurred during the
call. I traced the cause to the CDR on one of the channels being
NULL. murf suggested a check in the end bridge callback to be
sure the CDR is non-NULL before proceeding, so that's what I'm
adding. ........
2008-11-12 17:38 +0000 [r156166] Russell Bryant <russell@digium.com>
* /, main/asterisk.c: Merged revisions 156164 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r156164 | russell | 2008-11-12 11:29:52 -0600 (Wed, 12 Nov 2008)
| 7 lines Move the sanity check that makes sure "always fork" is
not set along with the console option to be after the code that
reads options from asterisk.conf. This resolves a situation where
Asterisk can start taking up 100% when misconfigured. (Thanks to
Bryce Porter (x86 on IRC) for letting me log in to his system to
figure out what was causing the 100% CPU problem.) ........
2008-11-12 17:28 +0000 [r156162] Eliel C. Sardanons <eliels@gmail.com>
* main/xmldoc.c: - The paramname is a pointer allocated with
strdup() or malloc(), so, we need to free it with ast_free().
2008-11-12 15:33 +0000 [r156127] Mark Michelson <mmichelson@digium.com>
* configure, autoconf/ast_c_compile_check.m4: Add a couple of
AC_SUBST calls to the AST_C_COMPILE_CHECK macro. These missing
calls were discovered when working on timerfd support in a
separate branch.
2008-11-12 13:43 +0000 [r156125] Eliel C. Sardanons <eliels@gmail.com>
* res/res_agi.c: Add XML documentation for AGI commands: - database
deltree - database get - exec - get data - get full variable
2008-11-12 06:46 +0000 [r156120] Michiel van Baak <michiel@vanbaak.info>
* main/udptl.c, main/pbx.c, channels/chan_sip.c,
configs/cli_aliases.conf.sample (added), include/asterisk/cli.h,
CHANGES, res/res_jabber.c, main/rtp.c, main/cli.c, main/cdr.c,
channels/chan_skinny.c, res/res_agi.c, pbx/pbx_ael.c,
pbx/pbx_dundi.c, funcs/func_devstate.c, main/asterisk.c,
channels/chan_mgcp.c, res/res_clialiases.c (added): This commit
does two things: - Add CLI aliases module to asterisk. - Remove
all deprecated CLI commands from the code Initial work done by
file. Junk-Y and lmadsen did a lot of work and testing to get the
list of deprecated commands into the configuration file.
Deprecated CLI commands are now handled by this new module, see
cli_aliases.conf for more info about that. ok russellb@ via
reviewboard (closes issue #13735) Reported by: mvanbaak
2008-11-12 02:20 +0000 [r156051-156087] Eliel C. Sardanons <eliels@gmail.com>
* res/res_agi.c, doc/appdocsxml.dtd: - Add 'database del',
'database put' and 'set music' AGI commands XML documentation. -
Add to the DTD the possibility to put a parameter inside an
<enum>.
* include/asterisk/agi.h, res/res_agi.c, doc/appdocsxml.dtd,
main/xmldoc.c: Implement AGI XML documentation parsing functions.
A new <agi> element is used to describe the XML documentation. We
have the usual synopsis,syntax,description and seealso for AGI
commands. The CLI 'agi show commands' command was changed to show
all the documentation se ctions.
2008-11-11 23:32 +0000 [r156017-156018] Pari Nannapaneni <paripurnachand@digium.com>
* main/manager.c: changing comment style to conform coding
guidelines
* main/manager.c: Patch by Ryan Brindley -- Make sure that manager
refuses any duplicate 'new category' requests in updateconfig
2008-11-11 17:57 +0000 [r155967] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk/strings.h: use some fancy compiler magic (thanks
to Matthew Woehlke on the gcc-help mailing list) to restore
type-safety to S_OR by going back to a macro, but preserve the
side-effect-safe usage of the macro arguments
2008-11-11 16:46 +0000 [r155934] Doug Bailey <dbailey@digium.com>
* res/res_phoneprov.c, phoneprov/polycom_line.xml: Add LINEKEYS
variable to allow for a user to set the number of keys assigned
to a line on a polycom phone
2008-11-11 16:07 +0000 [r155929] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Remove commentary from the issues list for
SIP TCP/TLS
2008-11-10 21:14 +0000 [r155863] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_agent.c: Merged revisions 155861 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r155861 | mmichelson | 2008-11-10 15:07:39 -0600 (Mon,
10 Nov 2008) | 14 lines Channel drivers assume that when their
indicate callback is invoked, that the channel on which the
callback was called is locked. This patch corrects an instance in
chan_agent where a channel's indicate callback is called directly
without first locking the channel. This was leading to some
observed locking issues in chan_local, but considering that all
channel drivers operate under the same expectations, the generic
fix in chan_agent is the right way to go. AST-126 ........
2008-11-10 21:12 +0000 [r155763-155862] Tilghman Lesher <tlesher@digium.com>
* res/res_realtime.c: Make documentation of update method match
documentation and update update2 method to match. Reported by:
atis, via -dev mailing list. Fixed by: me
* /, doc/valgrind.txt: Merged revisions 155803 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r155803 | tilghman | 2008-11-10 14:49:59 -0600 (Mon, 10 Nov 2008)
| 1 line I got tired of saying this in every single bugnote
referring to this file. ........
* main/editline/readline.c: Fix memory leak when MALLOC_DEBUG is
enabled. (closes issue #13864) Reported by: eliel Patches:
readline.c.patch uploaded by eliel (license 64)
2008-11-10 13:53 +0000 [r155711] Eliel C. Sardanons <eliels@gmail.com>
* main/pbx.c, main/Makefile, include/asterisk/xmldoc.h (added),
include/asterisk/term.h, include/asterisk/_private.h,
main/asterisk.c, main/xmldoc.c (added): Move all the XML
documentation API from pbx.c to xmldoc.c. Export the XML
documentation API: ast_xmldoc_build_synopsis()
ast_xmldoc_build_syntax() ast_xmldoc_build_description()
ast_xmldoc_build_seealso() ast_xmldoc_build_arguments()
ast_xmldoc_printable() ast_xmldoc_load_documentation()
2008-11-09 16:30 +0000 [r155554-155671] Sean Bright <sean.bright@gmail.com>
* configs/chan_dahdi.conf.sample: Fix this as well. Pointed out by
tzafrir.
* configs/chan_dahdi.conf.sample: Fix some spelling errors, and
convert tabs to spaces.
* main/channel.c, channels/chan_sip.c, apps/app_directed_pickup.c,
main/features.c, include/asterisk/channel.h: In order to move
away from nested function use, some changes to the recently
introduced ast_channel_search_locked need to be made.
Specifically, the caller needs to be able to pass arbitrary data
which in turn is passed to the callback. This patch addresses all
of the nested functions currently in asterisk trunk.
* apps/app_dial.c, /, main/features.c, include/asterisk/channel.h,
apps/app_followme.c, apps/app_queue.c: Merged revisions 155553
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r155553 | seanbright | 2008-11-08 20:08:07 -0500 (Sat, 08 Nov
2008) | 6 lines Use static functions here instead of nested ones.
This requires a small change to the ast_bridge_config struct as
well. To understand the reason for this change, see the following
post: http://gcc.gnu.org/ml/gcc-help/2008-11/msg00049.html
........
2008-11-08 21:46 +0000 [r155513-155516] Russell Bryant <russell@digium.com>
* channels/chan_sip.c, include/asterisk/strings.h: - Check for
failure when putting the packet in the ast_str - fix a spelling
error in a header file
* channels/chan_sip.c: Remove some code that is basically a no-op.
Code above this already ensures that the buffer is terminated.
2008-11-07 23:41 +0000 [r155467] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Set the invite state to INV_CANCELLED in a
place that makes more sense. Where it was set before, it was
impossible to actually delay sending a CANCEL if we had not yet
received a provisional response to an INVITE. (closes issue
#13626) Reported by: atis Patches: 13626.patch uploaded by
putnopvut (license 60) Tested by: atis
2008-11-07 22:39 +0000 [r155401] Sean Bright <sean.bright@gmail.com>
* main/manager.c, channels/chan_sip.c, funcs/func_dialgroup.c,
res/res_timing_pthread.c, include/asterisk/astobj2.h,
main/features.c, res/res_phoneprov.c, utils/hashtest2.c,
channels/chan_console.c, main/taskprocessor.c, apps/app_queue.c,
channels/chan_iax2.c, main/astobj2.c, main/config.c: Add ability
to pass arbitrary data to the ao2_callback_fn (called from
ao2_callback and ao2_find). Currently, passing OBJ_POINTER to
either of these mandates that the passed 'arg' is a hashable
object, making searching for an ao2 object based on outside
criteria difficult. Reviewed by Russell and Mark M. via
ReviewBoard: http://reviewboard.digium.com/r/36/
2008-11-07 22:28 +0000 [r155395-155399] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_sip.c: Merged revisions 155398 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r155398 | tilghman | 2008-11-07 16:27:32 -0600 (Fri, 07 Nov 2008)
| 7 lines Clarify error message. (closes issue #13809) Reported
by: denke Patches: 20081104__bug13809.diff.txt uploaded by
Corydon76 (license 14) Tested by: denke ........
* funcs/func_odbc.c: Two bugs relating to colnames found by
Marquis42 on #asterisk-dev
2008-11-07 21:14 +0000 [r155360] Mark Michelson <mmichelson@digium.com>
* configs/voicemail.conf.sample: Remove one more instance of the
sample configuration lying about what's possible. The tz cannot
be set in a context like this. It can only be set in the general
section or per-mailbox. Thanks to sasargen on #asterisk-dev for
pointing this out
2008-11-07 20:13 +0000 [r155324] Tilghman Lesher <tlesher@digium.com>
* channels/chan_dahdi.c: Send call release with unallocated cause
instead of normal call clearing, when invalid extension is
called. (closes issue #13408) Reported by: adomjan Patches:
chan_dahdi.c-ss7-unallocated-2 uploaded by adomjan (license 487)
2008-11-07 16:18 +0000 [r155284] Sean Bright <sean.bright@gmail.com>
* include/asterisk/indications.h, res/res_indications.c,
main/indications.c: Convert open-coded linked list in indications
to the AST_LIST_* macros. This cleans the code up some and should
make it more maintainable as time goes on. Reviewed by Russell,
Kevin, Mark M., and Tilghman via ReviewBoard:
http://reviewboard.digium.com/r/34/
2008-11-07 15:52 +0000 [r155282] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_sip.c: stringfields conversion for struct sip_peer,
as requested :-)
2008-11-07 15:42 +0000 [r155241-155264] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Remove a bogus ast_free() that Kevin
noticed. This was probably just left over from pre-astobj2ified
chan_sip.
* include/asterisk/astobj2.h: Clarify which part of OBJ_MULTIPLE is
not implemented, and under what case it is perfectly fine to use.
(Inspired by a question I received about my last commit.)
* main/pbx.c, channels/chan_sip.c: Fix some code in chan_sip that
was intended to unlink multiple objects from a container. The
OBJ_MULTIPLE flag must be provided here. Otherwise, this would
only remove a single object.
2008-11-07 03:09 +0000 [r155206] Kevin P. Fleming <kpfleming@digium.com>
* pbx/pbx_config.c: correct logic error noticed by rmudgett
(thanks!)
2008-11-07 03:02 +0000 [r155175-155204] Eliel C. Sardanons <eliels@gmail.com>
* main/pbx.c: If 'asterisk.conf' is not found, instead of giving
up, load documentation for the 'en_US' language (fix my last
commit).
* main/pbx.c: Fix an asterisk crash if no asterisk.conf
configuration file is present.
2008-11-06 22:49 +0000 [r155066-155121] Kevin P. Fleming <kpfleming@digium.com>
* res/ael/ael_lex.c, utils/extconf.c, res/ael/ael.flex: don't
blindly assume that Darwin and Cygwin need GLOB_ABORTED defined;
only define it if it is not already defined
* pbx/pbx_config.c: coding style/guidelines cleanup, plus use new
side-effect safe S_OR
* include/asterisk/strings.h: make S_OR and S_COR safe to use even
if the parameters are function calls or have side effects. it
still bothers me that these are called S_OR and not something
like ast_string_or, but that's water over the bridge
* channels/chan_dahdi.c: put ifdef protection around the rest of
the libpri function calls that were added at the same time as
progress_with_cause move parsing of the qsig channel mapping
configuration option outside ifdef HAVE_PRI_INBANDDISCONNECT and
into a properly ifdef'd block
2008-11-06 19:46 +0000 [r155012] Mark Michelson <mmichelson@digium.com>
* /, configs/voicemail.conf.sample: Merged revisions 155011 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r155011 | mmichelson | 2008-11-06 13:45:52 -0600 (Thu, 06 Nov
2008) | 8 lines The documentation listed the ability to set
'maxmsg' per context. The truth is that you can only set this in
the general section or per mailbox. Thus I am updating the sample
config file to be more accurate. Thanks to sasargen on IRC for
bringing up this issue. ........
2008-11-06 18:19 +0000 [r154967] Eliel C. Sardanons <eliels@gmail.com>
* main/pbx.c: Simplify the output of [See Also]. Functions are
printed without parenthesis like: FUNTION Applications are
printed with parenthesis like: AppName() Cli commands are printed
like: 'core show application' The other type of references are
printed as they are inside the <ref> tag.
2008-11-05 22:22 +0000 [r154923-154926] Sean Bright <sean.bright@gmail.com>
* apps/app_directed_pickup.c: Fix some whitespace.
* apps/app_directed_pickup.c, main/features.c: Update a couple
places to use the new ast_channel_search_locked API call.
2008-11-05 22:19 +0000 [r154922] Tilghman Lesher <tlesher@digium.com>
* main/asterisk.c: Don't read history on -rx commands. (Closes
issue #13571) Reported by: tzafrir Patch
'0001-no-need-for-history-on-asterisk-rx.patch' uploaded by
tzafrir.
2008-11-05 22:01 +0000 [r154919] Sean Bright <sean.bright@gmail.com>
* include/asterisk.h: Fix a problem found while building res_snmp.
2008-11-05 21:58 +0000 [r154915] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/app.h, funcs/func_strings.c, main/app.c,
CHANGES: Add LISTFILTER dialplan function, along with supporting
documentation. See documentation for more information on how to
use it.
2008-11-05 20:45 +0000 [r154875] Matthew Fredrickson <creslin@digium.com>
* channels/chan_dahdi.c, configure,
include/asterisk/autoconfig.h.in, configure.ac: Make compilation
of chan_dahdi so that it does not require the new
pri_progress_with_cause function to have libpri support work.
2008-11-05 20:33 +0000 [r154839] Michiel van Baak <michiel@vanbaak.info>
* res/res_http_post.c: make this compile on OpenBSD again.
2008-11-05 20:17 +0000 [r154796-154837] Eliel C. Sardanons <eliels@gmail.com>
* channels/chan_agent.c: Add AgentLogin(), AgentMonitorOutgoing()
applications and AGENT() function XML documentation.
* apps/app_test.c: Add TestClient() and TestServer() applications
XML documentation.
* apps/app_mixmonitor.c: Add more [see also] references based on
TFOT.
* apps/app_macro.c: Add Macro(), MacroExit(), MacroExclusive() and
MacroIf() applications XML documentation. (closes issue #13699)
Reported by: snuffy Patches: bug13699_20081016.diff uploaded by
snuffy (license 35)
2008-11-05 16:11 +0000 [r154687] Steve Murphy <murf@digium.com>
* main/channel.c, /: Merged revisions 154685 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r154685 | murf | 2008-11-05 09:06:53 -0700 (Wed, 05 Nov 2008) | 1
line This fix was prompted by communication from user, who was
seeing thousands of error logs... looks like EAGAIN. Made such
uninteresting. ........
2008-11-05 14:37 +0000 [r154467-154647] Eliel C. Sardanons <eliels@gmail.com>
* main/pbx.c, apps/app_privacy.c, apps/app_sayunixtime.c,
main/features.c, apps/app_morsecode.c, apps/app_alarmreceiver.c,
apps/app_amd.c: Add more SeeAlso references based on TFOT.
* doc/appdocsxml.dtd: We now can have a reference to a filename
inside a <see-also> tag.
* apps/app_parkandannounce.c: - Add ParkAndAnnounce() application
XML documentation.
* main/pbx.c, apps/app_page.c, apps/app_authenticate.c,
apps/app_dumpchan.c, apps/app_disa.c, apps/app_image.c,
apps/app_chanspy.c, apps/app_stack.c, apps/app_adsiprog.c: - Add
more <see-also> based on TFOT. - Add the 'filename' type to the
see-also ref. To be able to reference a filename.
* apps/app_readfile.c, funcs/func_db.c, apps/app_sendtext.c,
funcs/func_blacklist.c, apps/app_url.c, apps/app_queue.c,
apps/app_senddtmf.c, apps/app_db.c: - Add some see-also
references based on TFOT.
* apps/app_read.c: - Add Read() application XML documentation.
* apps/app_followme.c: - Add FollowMe() application XML
documentation.
* apps/app_forkcdr.c, res/res_indications.c: - Add PlayTones() and
StopPlayTones() applications XML documentation. - Fix a dot that
was outside of the <para> in the ForkCDR() XML documentation.
2008-11-04 23:23 +0000 [r154429] Sean Bright <sean.bright@gmail.com>
* main/channel.c, channels/chan_sip.c, include/asterisk/channel.h:
Introduce a new API call ast_channel_search_locked, which
iterates through the channel list calling a caller-defined
callback. The callback returns non-zero if a match is found. This
should speed up some of the code that I committed earlier today
in chan_sip (which is also updated by this commit). Reviewed by
russellb and kpfleming via ReviewBoard:
http://reviewboard.digium.com/r/28/
2008-11-04 23:03 +0000 [r154366-154428] Tilghman Lesher <tlesher@digium.com>
* channels/chan_iax2.c: Switch to using a thread condition to
signal that a child thread is ready for work, rather than a busy
wait. (closes issue #13011) Reported by: jpgrayson Patches:
chan_iax2_find_idle.patch uploaded by jpgrayson (license 492)
* /, channels/chan_iax2.c: Merged revisions 154365 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r154365 | tilghman | 2008-11-04 14:49:33 -0600 (Tue, 04
Nov 2008) | 9 lines On busy systems, it's possible for the values
checked within a single line of code to change, unless the
structure is locked to ensure a consistent state. (closes issue
#13717) Reported by: kowalma Patches: 20081102__bug13717.diff.txt
uploaded by Corydon76 (license 14) Tested by: kowalma ........
2008-11-04 20:12 +0000 [r154329] Eliel C. Sardanons <eliels@gmail.com>
* Makefile: We need to pass the DTD to xmlstarlet to validate
against it the XML. (I thought it was being read within the
DOCTYPE definition inside the XML).
2008-11-04 19:07 +0000 [r154268] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c, /: Merged revisions 154266 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r154266 | rmudgett | 2008-11-04 13:01:08 -0600 (Tue, 04
Nov 2008) | 4 lines JIRA ABE-1703 mISDN sets the channel to the
wrong state when it receives the indication AST_CONTROL_RINGING.
........
2008-11-04 18:59 +0000 [r154260-154264] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_skinny.c, channels/chan_h323.c: Recorded merge
of revisions 154263 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r154263 | tilghman | 2008-11-04 12:58:05 -0600 (Tue, 04 Nov 2008)
| 3 lines Make the monitor thread non-detached, so it can be
joined (suggested by Russell on -dev list). ........
* include/asterisk/devicestate.h, main/manager.c, apps/app_page.c,
include/asterisk/config.h, main/features.c, main/devicestate.c,
apps/app_queue.c, main/config.c, apps/app_voicemail.c: Slightly
optimize ast_devstate_str and rename global functions
devstate2str and config_text_file_save to have an ast_ prefix
2008-11-04 18:06 +0000 [r154225] Eliel C. Sardanons <eliels@gmail.com>
* apps/app_forkcdr.c: Add XML documentation for the ForkCDR()
application.
2008-11-04 17:23 +0000 [r154186-154191] Sean Bright <sean.bright@gmail.com>
* main/pbx.c: GLOB_BRACE is already added to MY_GLOB_FLAGS if it is
supported on the platform. This should resolve some build errors
on Solaris. (issue #13704) Reported by: dougm
* channels/chan_sip.c, configs/sip.conf.sample: Allow devices that
accept dialog-info+xml (like snoms) to get the Caller ID of the
calling party when subscribed to the state of an extension that
is ringing. This has some limitations which are documented in
sip.conf.sample. (closes issue #13827) Reported by: seanbright
Patches: issue13827.patch uploaded by seanbright (license 71)
Reviewed by: russellb
* main/Makefile: Fix build errors.
2008-11-04 15:07 +0000 [r154151] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_vpb.cc, res/res_crypto.c, configure.ac,
cdr/cdr_adaptive_odbc.c, channels/chan_oss.c,
channels/chan_usbradio.c, res/res_config_odbc.c,
apps/app_osplookup.c, funcs/func_odbc.c, configure,
build_tools/menuselect-deps.in, channels/chan_alsa.c,
makeopts.in, cdr/cdr_odbc.c, res/res_odbc.c,
apps/app_voicemail.c: improve configure script to remember the
previous value of each dependency in build_tools/menuselect-deps,
so that (once it has been written) menuselect can use this
information to warn the user when a previously met dependency is
no longer met along the way, change tags used in configure
script, menuselect-deps and code for various dependencies to be
consistently named
2008-11-04 14:38 +0000 [r154149] Eliel C. Sardanons <eliels@gmail.com>
* channels/chan_dahdi.c: Add XML documentation for: Applications -
DAHDISendKeypadFacility() - DAHDISendCallreroutingFacility()
2008-11-03 22:28 +0000 [r154023-154072] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_voicemail.c: Merged revisions 154066 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r154066 | tilghman | 2008-11-03 16:27:10 -0600 (Mon, 03
Nov 2008) | 5 lines Attempting to expunge a mailbox when the
mailstream is NULL will crash Asterisk. (Closes issue #13829)
Reported by: jaroth Patch by: me (modified jaroth's patch)
........
* /, main/rtp.c: Merged revisions 154060 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r154060 | tilghman | 2008-11-03 15:48:21 -0600 (Mon, 03 Nov 2008)
| 3 lines Remove the potential for a division by zero error.
(Closes issue #13810) ........
* funcs/func_odbc.c: Should have passed the string pointer, not the
ast_str structure. (closes issue #13830) Reported by: Marquis
2008-11-03 18:02 +0000 [r153983] Olle Johansson <oej@edvina.net>
* configs/sip.conf.sample: Updating docs
2008-11-03 17:11 +0000 [r153947] Eliel C. Sardanons <eliels@gmail.com>
* apps/app_stack.c: Add LOCAL() function XML documentation.
2008-11-03 15:25 +0000 [r153904-153905] Olle Johansson <oej@edvina.net>
* configs/sip.conf.sample: Spaces to replace tabs...
* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Adding a
separation of remote authentication and our authentication.
remotesecret => our password for a remote service secret => our
authentication when someone calls us Secret => still has both
functions if remotesecret is not used.
2008-11-03 13:33 +0000 [r153803-153852] Eliel C. Sardanons <eliels@gmail.com>
* channels/chan_iax2.c: Add XML documentation for: Functions -
IAXPEER() - IAXVAR()
* channels/chan_sip.c: Add XML documentation for: Applications -
SIPDtmfMode() - SIPAddHeader() Functions - SIP_HEADER() -
SIPPEER() - SIPCHANINFO() - CHECKSIPDOMAIN()
2008-11-03 12:26 +0000 [r153787] Kevin P. Fleming <kpfleming@digium.com>
* configure, autoconf/ast_ext_lib.m4: when --without-<foo> is
passed to the configure script, explicitly inform menuselect that
the package was disabled by the user
2008-11-03 01:01 +0000 [r153747] Eliel C. Sardanons <eliels@gmail.com>
* apps/app_waitforring.c, apps/app_waitforsilence.c, apps/app_db.c,
apps/app_ivrdemo.c: Add XML documentation for: - WaitForSilence()
- WaitForNoise() - WaitForRing() - IVRDemo() - DBDel() -
DBDeltree() (issue #13699) Reported by: snuffy Patches:
bug13699_20081016.diff uploaded by snuffy (license 35) (With
minor changes)
2008-11-02 23:34 +0000 [r153709] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk/agi.h, configure,
include/asterisk/autoconfig.h.in, autoconf/ast_gcc_attribute.m4,
configure.ac, include/asterisk/compiler.h, apps/app_stack.c:
instead of trying to forcibly load res_agi when app_stack is
loaded (even if the administrator didn't want it loaded), use GCC
weak symbols to determine whether it was loaded already or not;
if it was loaded, then use it.
2008-11-02 20:06 +0000 [r153652] Russell Bryant <russell@digium.com>
* /, include/asterisk/features.h: Merged revisions 153651 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r153651 | russell | 2008-11-02 13:51:17 -0600 (Sun, 02 Nov 2008)
| 2 lines features.h depends on linkedlists.h, so include it
........
2008-11-02 19:39 +0000 [r153616-153650] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_dahdi.c: fix one more warning missed because i did
not have new enough libpri installed
* res/res_musiconhold.c: fix small bug introduced while cleaning up
compiler warnings
* /: mark this revision as merged manually
* utils/muted.c, apps/app_authenticate.c, res/res_phoneprov.c,
main/utils.c, formats/format_wav_gsm.c, res/res_http_post.c,
res/res_musiconhold.c, channels/chan_iax2.c, res/res_jabber.c,
res/res_config_sqlite.c, utils/frame.c, utils/stereorize.c,
main/channel.c, channels/chan_dahdi.c, main/manager.c,
res/ael/ael.tab.c, funcs/func_odbc.c, main/ast_expr2f.c,
res/res_agi.c, main/http.c, main/logger.c, formats/format_gsm.c,
apps/app_adsiprog.c, apps/app_dial.c, channels/chan_sip.c,
apps/app_festival.c, formats/format_wav.c, res/ael/ael.y,
main/db1-ast/hash/hash_page.c, agi/eagi-test.c, res/res_crypto.c,
utils/astman.c, pbx/pbx_lua.c, formats/format_ogg_vorbis.c,
utils/astcanary.c, apps/app_queue.c, channels/chan_oss.c,
agi/eagi-sphinx-test.c, res/ael/ael_lex.c, channels/chan_h323.c,
main/file.c, apps/app_sms.c, pbx/pbx_dundi.c, res/ael/ael.flex,
pbx/pbx_config.c, apps/app_chanspy.c, apps/app_stack.c,
utils/streamplayer.c, main/asterisk.c, apps/app_voicemail.c:
bring over all the fixes for the warnings found by gcc 4.3.x from
the 1.4 branch, and add the ones needed for all the new code here
too
2008-11-02 06:24 +0000 [r153582] Eliel C. Sardanons <eliels@gmail.com>
* channels/chan_iax2.c: Add IAX2Provision() application XML
documentation.
2008-11-02 05:56 +0000 [r153577-153580] Russell Bryant <russell@digium.com>
* Makefile: validate-docs is a PHONY target
* Makefile, configure, configure.ac, makeopts.in: Add a handy
makefile target so that you can validate the documentation
against the DTD by running "make validate-docs"
* Makefile: Modify the Makefile logic for extracting documentation.
- Build the documentation when you run "make", as opposed to
"make install" - Only rebuild the documentation when source code
has been changed
2008-11-02 05:10 +0000 [r153541-153543] Eliel C. Sardanons <eliels@gmail.com>
* apps/app_flash.c: Add Flash() application XML documentation.
* apps/app_talkdetect.c: Fix a typo in the name of the application.
2008-11-02 04:14 +0000 [r153472-153507] Sean Bright <sean.bright@gmail.com>
* channels/Makefile: There is a troublesome assert() in the
alsa/control.h header that causes GCC 4.3.2 to complain that the
passed argument will always evaluate to true. So to get things to
compile, disable assert when building chan_usbradio.so.
* apps/app_record.c: Another little one.
2008-11-02 02:55 +0000 [r153362-153470] Russell Bryant <russell@digium.com>
* apps/app_page.c: fix a typo (thanks sean)
* apps/app_dial.c, funcs/func_speex.c, apps/app_page.c,
apps/app_record.c, funcs/func_env.c, apps/app_dahdiras.c,
funcs/func_math.c, funcs/func_strings.c, apps/app_userevent.c,
apps/app_exec.c, apps/app_chanspy.c, apps/app_playback.c: Fix
various spelling and grammatical issues in documentation
* apps/app_voicemail.c: - Use a for loop instead of a while loop -
Get rid of an unnecessary variable
* apps/app_directed_pickup.c: Instead of doing a couple of strlen()
calls each iteration of the loop, only do it once at the
beginning of the function
* channels/chan_sip.c: Don't ignore the result of find_peer() when
looking for a peer by IP in check_peer_ok().
* funcs/func_speex.c, apps/app_dahdibarge.c, funcs/func_rand.c,
apps/app_readfile.c, funcs/func_module.c, funcs/func_dialgroup.c,
include/asterisk/autoconfig.h.in, funcs/func_env.c,
apps/app_dahdiscan.c, apps/app_record.c, funcs/func_strings.c,
apps/app_sayunixtime.c, include/asterisk/extconf.h,
apps/app_alarmreceiver.c, apps/app_image.c,
apps/app_chanisavail.c, apps/app_ices.c, apps/app_exec.c,
main/config.c, main/term.c, include/asterisk/compat.h, configure,
funcs/func_shell.c, apps/app_skel.c, apps/app_dumpchan.c,
include/asterisk/module.h, main/features.c, apps/app_amd.c,
apps/app_url.c, apps/app_milliwatt.c, apps/app_dial.c,
main/pbx.c, include/asterisk/xml.h (added), apps/app_page.c,
funcs/func_timeout.c, main/Makefile, apps/app_privacy.c,
apps/app_echo.c, apps/app_softhangup.c, apps/app_fax.c,
funcs/func_math.c, apps/app_dahdiras.c, configure.ac,
apps/app_disa.c, apps/app_morsecode.c, funcs/func_cut.c,
apps/app_talkdetect.c, apps/app_transfer.c, apps/app_playback.c,
doc/tex/asterisk-conf.tex, Makefile, apps/app_sendtext.c,
funcs/func_channel.c, funcs/func_cdr.c, apps/app_zapateller.c,
build_tools/get_documentation (added), funcs/func_iconv.c,
apps/app_mixmonitor.c, apps/app_chanspy.c, main/asterisk.c,
apps/app_cdr.c, funcs/func_base64.c, funcs/func_md5.c,
apps/app_dictate.c, apps/app_authenticate.c,
apps/app_readexten.c, apps/app_userevent.c, funcs/func_vmcount.c,
main/xml.c (added), funcs/func_sha1.c, funcs/func_logic.c,
funcs/func_uri.c, apps/app_controlplayback.c, funcs/func_enum.c,
apps/app_setcallerid.c, funcs/func_groupcount.c,
funcs/func_config.c, funcs/func_volume.c, funcs/func_odbc.c,
apps/app_mp3.c, apps/app_directory.c, apps/app_jack.c,
apps/app_adsiprog.c, apps/app_while.c, apps/app_nbscat.c,
funcs/func_dialplan.c, funcs/func_db.c, funcs/func_version.c,
apps/app_festival.c, funcs/func_lock.c, apps/app_waituntil.c,
doc, include/asterisk/term.h, include/asterisk/_private.h,
apps/app_system.c, apps/app_getcpeid.c, apps/app_queue.c,
funcs/func_global.c, funcs/func_extstate.c,
funcs/func_realtime.c, apps/app_channelredirect.c,
funcs/func_blacklist.c, apps/app_directed_pickup.c,
include/asterisk/pbx.h, include/asterisk/strings.h, makeopts.in,
apps/app_senddtmf.c, funcs/func_devstate.c,
funcs/func_callerid.c, doc/appdocsxml.dtd (added),
apps/app_verbose.c, apps/app_stack.c: Merge changes from
team/group/appdocsxml This commit introduces the first phase of
an effort to manage documentation of the interfaces in Asterisk
in an XML format. Currently, a new format is available for
applications and dialplan functions. A good number of conversions
to the new format are also included. For more information, see
the following message to asterisk-dev:
http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html
* channels/chan_sip.c: Ensure that the sip_pvt properly has its
refcount incremented when the scheduler holds a reference to it
for session timer processing.
2008-11-01 01:55 +0000 [r153296] Sean Bright <sean.bright@gmail.com>
* configs/sip.conf.sample: The default in chan_sip for
notifyringing is yes, so update the sample conf to reflect that.
2008-10-31 20:05 +0000 [r153223] Mark Michelson <mmichelson@digium.com>
* main/dial.c, apps/app_page.c, include/asterisk/dial.h, CHANGES: *
Fixed timeout logic in the dialing API as setting timeouts had no
effect * Updated dialing API documentation to indicate that
timeouts are specified in milliseconds * Added a new timeout
argument to the Page application. If time expires, any endpoints
which have not answered will be hung up.
2008-10-31 18:55 +0000 [r153181] Terry Wilson <twilson@digium.com>
* apps/app_dial.c, main/features.c, include/asterisk/channel.h,
apps/app_followme.c, apps/app_queue.c: Recent CDR fixes moved
execution of the 'h' exten into the bridging code, so variables
that were set after ast_bridge_call was called would not show up
in the 'h' exten. Added a callback function to handle setting
variables, etc. from w/in the bridging code. Calls back into a
nested function within the function calling ast_bridge_call
(closes issue #13793) Reported by: greenfieldtech
2008-10-31 17:18 +0000 [r153122-153124] Tilghman Lesher <tlesher@digium.com>
* configs/func_odbc.conf.sample, funcs/func_odbc.c, CHANGES:
Failover for func_odbc, allowing an INSERT query to be performed
when the UPDATE query initially affects 0 rows. (closes issue
#13083) Reported by: Corydon76 Patches:
20081031__bug13083.diff.txt uploaded by Corydon76 (license 14)
* /, channels/chan_sip.c: Merged revisions 153114 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r153114 | tilghman | 2008-10-31 11:30:32 -0500 (Fri, 31 Oct 2008)
| 3 lines Turn off qualify on uncached realtime peers. (Closes
issue #13383) ........
2008-10-31 09:31 +0000 [r153057] Russell Bryant <russell@digium.com>
* main/channel.c: Use the ast_str API call to reset the string
instead of manually editing its internals (closes issue #13816)
Reported by: eliel Patches: channel.c.patch uploaded by eliel
(license 64)
2008-10-30 20:59 +0000 [r152993] Sean Bright <sean.bright@gmail.com>
* /, bootstrap.sh: Merged revisions 152992 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r152992 | seanbright | 2008-10-30 16:58:24 -0400 (Thu, 30 Oct
2008) | 2 lines The -I argument to aclocal needs a space before
the include directory name. ........
2008-10-30 20:46 +0000 [r152990] Russell Bryant <russell@digium.com>
* include/asterisk/timing.h: Add a todo for a new timing API
implementation that would work for Linux systems as of kernel
2.6.25 and glibc 2.8
2008-10-30 20:35 +0000 [r152923-152969] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_h323.c: Merged revisions 152958 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r152958 | tilghman | 2008-10-30 15:33:28 -0500 (Thu, 30
Oct 2008) | 3 lines Cannot join detached threads. See
http://www.opengroup.org/onlinepubs/000095399/functions/pthread_join.html
(Closes issue #13400) ........
* channels/chan_local.c, /: Merged revisions 152922 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r152922 | tilghman | 2008-10-30 14:43:38 -0500 (Thu, 30
Oct 2008) | 6 lines Unlock before returning, when extension
doesn't exist. (closes issue #13807) Reported by: eliel Patches:
chan_local.c.patch uploaded by eliel (license 64) ........
2008-10-30 19:40 +0000 [r152887-152920] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Fix the sip_peer reference count with
respect to scheduler entries for scheduling peer pokes, and
scheduling peer poke expirations.
* channels/chan_sip.c: Fix the sip_peer reference count with
respect to scheduler entries for registration expirations.
* include/asterisk/sched.h: Fix a bug in AST_SCHED_REPLACE_UNREF().
The reference count of the object _must_ be increased before
creating the scheduler entry. Otherwise, you create a race
condition where the reference count may hit zero and the object
can disappear out from under you. This could also would have
incorrectly decreased the reference count in the case that the
scheduler add failed.
2008-10-30 19:23 +0000 [r152879] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: I just noticed this construct and thought it
was silly to have a bunch of case statements with duplicated code
in each case. Instead, just use the built-in fallthrough
capability of case statements and reduce the code to a single
instance
2008-10-30 19:21 +0000 [r152875-152877] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Modify the documentation of the sip_registry
struct - Remove a comment that says that the monitor thread is
the only one that ever touches these objects. This is no longer
the case with TCP. Also, I would eventually like to get the
scheduler in its own thread, so this is just a poor assumption to
make. - Note that reference counting of these objects with
respect to scheduler entries is not complete. There are some
leaked references when deleting scheduler entries.
* funcs/func_db.c: - spaces to tabs - add some braces - remove
unnecessary cast
2008-10-30 16:54 +0000 [r152809-152812] Kevin P. Fleming <kpfleming@digium.com>
* main/cdr.c, /: Merged revisions 152811 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r152811 | kpfleming | 2008-10-30 11:53:48 -0500 (Thu, 30 Oct
2008) | 3 lines instead of comparing the string pointer to 0,
let's compare the value that was actually parsed out of the
string (found by sparse) ........
* include/asterisk/buildinfo.h (added): try to get this committed
before the buildbot complains about a broken tree
* channels/misdn/isdn_lib_intern.h, channels/misdn/isdn_lib.h,
main/dial.c, main/dnsmgr.c, main/buildinfo.c,
codecs/lpc10/chanwr.c, utils/astcanary.c,
channels/misdn/isdn_lib.c, main/asterisk.c, apps/app_adsiprog.c:
fix a few small things found by using sparse
2008-10-30 16:38 +0000 [r152807] Mark Michelson <mmichelson@digium.com>
* main/features.c, CHANGES, configs/features.conf.sample: After
seeing another problem in #asterisk stemming from the low default
value of featuredigittimeout, I decided it was high time to
change it. I have changed the default to 2000 ms based on a
suggestion from Leif Madsen.
2008-10-30 04:26 +0000 [r152689-152765] Tilghman Lesher <tlesher@digium.com>
* configs/extensions.conf.sample: Set up an example stdexten that
preserves the original context and extension in the CDR. (Related
to issue #13799) Reported by: davidw
* CHANGES, apps/app_directory.c: Pay attention to the
searchcontexts entry in voicemail.conf (related to AST-125)
* main/pbx.c: Track down and fix annoying lock errors
2008-10-29 20:53 +0000 [r152646] Mark Michelson <mmichelson@digium.com>
* apps/app_directory.c: If there was no named defined in a
voicemail.conf mailbox entry, then app_directory would crash when
attempting to read that entry from the file. We now check for the
NULL or empty string properly so that there will be no crash.
(closes issue #13804) Reported by: bluecrow76
2008-10-29 05:47 +0000 [r152605] Steve Murphy <murf@digium.com>
* apps/app_dial.c, /, apps/app_queue.c,
configs/features.conf.sample: Merged revisions 152538 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r152538 | murf | 2008-10-28 23:19:04 -0600 (Tue, 28 Oct 2008) |
14 lines A little documentation cross-ref between features and
dial and queue... I wasted some time (stupidly) trying to get the
one-touch parking stuff working, because it didn't occur to me
that I had to also have the corresponding options in the dial
command! Duh! (In all this time, I never set this up before!) So,
to keep some poor fool from suffering the same fate, I made the
features.conf.sample file mention the corresponding opts in
dial/queue; and the docs for dial/app specifically mention the
corresponding decls in the feature.conf file. I hope this doesn't
spoil some vast, eternal plan... ........
2008-10-29 05:34 +0000 [r152569] Russell Bryant <russell@digium.com>
* /, channels/chan_sip.c: Merged revisions 152539 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r152539 | russell | 2008-10-29 00:23:51 -0500 (Wed, 29 Oct 2008)
| 7 lines Fix an incorrect usage of sizeof() (closes issue
#13795) Reported by: andrew53 Patches: chan_sip_sizeof.patch
uploaded by andrew53 (license 519) ........
2008-10-29 05:01 +0000 [r152536] Steve Murphy <murf@digium.com>
* apps/app_dial.c, /, main/features.c, include/asterisk/pbx.h,
apps/app_queue.c, include/asterisk/features.h: Merged revisions
152535 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r152535 | murf | 2008-10-28 22:36:32 -0600 (Tue, 28 Oct 2008) |
46 lines The magic trick to avoid this crash is not to try to
find the channel by name in the list, which is slow and resource
consuming, but rather to pay attention to the result codes from
the ast_bridge_call, to which I added the
AST_PBX_NO_HANGUP_PEER_PARKED value, which now are returned when
a channel is parked. Why? because CDR's aren't generated via
parking, so nothing is needed, but if a transfer occurred, there
are critical things I need. If you get AST_PBX_KEEPALIVE, then
don't touch the channel pointer. If you get
AST_PBX_NO_HANGUP_PEER, or AST_PBX_NO_HANGUP_PEER_PARKED, then
don't touch the peer pointer. Updated the several places where
the results from a bridge were not being properly obeyed, and
fixed some code I had introduced so that the results of the
bridge were not overridden (in trunk). All the places that
previously tested for AST_PBX_NO_HANGUP_PEER now have to check
for both AST_PBX_NO_HANGUP_PEER and
AST_PBX_NO_HANGUP_PEER_PARKED. I tested this against the 4 common
parking scenarios: 1. A calls B; B answers; A parks B; B hangs up
while A is getting the parking slot announcement, immediately
after being put on hold. 2. A calls B; B answers; A parks B; B
hangs up after A has been hung up, but before the park times out.
3. A calls B; B answers; B parks A; A hangs up while B is getting
the parking slot announcement, immediately after being put on
hold. 4. A calls B; B answers; B parks A; A hangs up after B has
been hung up, but before the park times out. No crash. I also ran
the scenarios above against valgrind, and accesses looked good.
........
2008-10-28 22:33 +0000 [r152467] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_voicemail.c: Merged revisions 152463 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r152463 | tilghman | 2008-10-28 17:32:34 -0500 (Tue, 28
Oct 2008) | 3 lines Quoting in the wrong direction (Fixes
AST-107) ........
2008-10-28 22:26 +0000 [r152448] Doug Bailey <dbailey@digium.com>
* configs/phoneprov.conf.sample: Add more polycom firmware files to
static mapping
2008-10-28 21:38 +0000 [r152369-152442] Tilghman Lesher <tlesher@digium.com>
* channels/chan_mgcp.c: Only re-add the io port if it was closed,
otherwise reload causes a memory leak. (closes issue #13785)
Reported by: eliel Patches: chan_mgcp.c.patch uploaded by eliel
(license 64)
* apps/app_dial.c, /: Merged revisions 152368 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r152368 | tilghman | 2008-10-28 12:04:56 -0500 (Tue, 28 Oct 2008)
| 8 lines Reset all DIAL variables back to blank, in case Dial is
called multiple times per call (which could otherwise lead to
inconsistent status reports). (closes issue #13216) Reported by:
ruddy Patches: 20081014__bug13216.diff.txt uploaded by Corydon76
(license 14) Tested by: ruddy ........
2008-10-27 23:31 +0000 [r152287] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 152286 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r152286 | jpeeler | 2008-10-27 18:28:49 -0500 (Mon, 27
Oct 2008) | 2 lines Buffer policy setting for half is not needed.
........
2008-10-27 21:34 +0000 [r152134-152216] Tilghman Lesher <tlesher@digium.com>
* channels/chan_local.c, /: Merged revisions 152215 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r152215 | tilghman | 2008-10-27 16:32:00 -0500 (Mon, 27
Oct 2008) | 6 lines Inherit ALL elements of CallerID across a
local channel. (closes issue #13368) Reported by: Peter Schlaile
Patches: 20080826__bug13368.diff.txt uploaded by Corydon76
(license 14) ........
* apps/app_stack.c: Set ARGC in subroutines with the number of
arguments passed.
* apps/app_stack.c: Oops, only delete the ARG variables once upon
release. The following section would have removed them again
(removing variables from 2 stack frames, instead of just one).
2008-10-27 16:03 +0000 [r152132] Jason Parker <jparker@digium.com>
* apps/app_transfer.c: Remove options argument parsing/syntax (it
isn't used any longer) (closes issue #13789) Reported by: IgorG
Patches: app_transfer.c.diff uploaded by IgorG (license 20)
2008-10-26 20:25 +0000 [r152060] Sean Bright <sean.bright@gmail.com>
* /, funcs/func_strings.c: Merged revisions 152059 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r152059 | seanbright | 2008-10-26 16:23:36 -0400 (Sun,
26 Oct 2008) | 7 lines Since passing \0 as the second argument to
strchr is valid (and will match the trailing \0 of a string) we
need to check that first, otherwise we end up with incorrect
results. Fix suggested by reporter. (closes issue #13787)
Reported by: meitinger ........
2008-10-26 10:23 +0000 [r151980-152020] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Trying to fix the user/peer matching
correctly. This will need some testing before getting merged into
1.6.1
* channels/chan_sip.c: Moving more variables to the sip_cfg
structure, as I have some future ideas for the usage of that
structure.
* channels/chan_sip.c: Doxygen changes and some formatting.
2008-10-25 11:02 +0000 [r151906] Russell Bryant <russell@digium.com>
* /, main/asterisk.c: Merged revisions 151905 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r151905 | russell | 2008-10-25 05:59:02 -0500 (Sat, 25 Oct 2008)
| 8 lines Move AMI initialization to occur after loading modules.
This prevents a deadlock when someone tries to initiate a module
reload from the AMI just as Asterisk is starting. (closes issue
#13778) Reported by: hotsblanc Fix suggested by hotsblanc
........
2008-10-23 21:27 +0000 [r151830] Terry Wilson <twilson@digium.com>
* funcs/func_odbc.c: allow to compile under --enable-dev-mode (gcc
didn't actually complain when I was using ccache...)
2008-10-23 15:54 +0000 [r151762] Tilghman Lesher <tlesher@digium.com>
* contrib/scripts/vmdb.sql: Clarify documentation, following merge
of realtime_update2 branch
2008-10-23 15:38 +0000 [r151739-151761] Olle Johansson <oej@edvina.net>
* CHANGES: Thanks russellb for reminding an old man....
* channels/chan_sip.c, doc/tex/channelvariables.tex: Adding a small
new feature. Setting _SIPFROMDOMAIN in a channel will set the
domain we use for the URI in the outbound call leg.
2008-10-23 15:28 +0000 [r151732] Tilghman Lesher <tlesher@digium.com>
* funcs/func_odbc.c: Simplify some nested functions, as suggested
by Russell on -dev
2008-10-23 15:09 +0000 [r151722] Doug Bailey <dbailey@digium.com>
* res/res_http_post.c: Add patch to handle how IE7 issues POST
requests using Window path spec including backslash delimiters
2008-10-22 22:11 +0000 [r151682] Tilghman Lesher <tlesher@digium.com>
* funcs/func_odbc.c, CHANGES: Added debugging CLI functions
2008-10-22 20:45 +0000 [r151642] BJ Weschke <bweschke@btwtech.com>
* channels/chan_sip.c: revert the changes in issue #13705 - it's
being re-opened as while the results fixed the complaint in the
issue, it introduced other more undesirable issues than what was
already reported
2008-10-22 20:05 +0000 [r151601] Tilghman Lesher <tlesher@digium.com>
* contrib/scripts/live_ast (added): Add a contributed script for
running Asterisk without installing it, first. (closes issue
#11680) Reported by: tzafrir Patches: live_ast_6 uploaded by
tzafrir (license 46)
2008-10-22 20:05 +0000 [r151600] Mark Michelson <mmichelson@digium.com>
* channels/chan_dahdi.c: Change some logical ands to bitwise ands
and add messages alerting that a channel is being ignored if the
PROC_DAHDI_NOCHAN option is set in process_dahdi. (closes issue
#13759) Reported by: smurfix Patches: dahdi.patch uploaded by
smurfix (license 547)
2008-10-22 17:45 +0000 [r151554-151555] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Print out the right var in the log message
* channels/chan_sip.c: Fix this check to use the proper variable
(the result from get_in_brackets)
2008-10-22 15:08 +0000 [r151420-151512] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: The logic of a strncasecmp call was
reversed. (closes issue #13706) Reported by: andrew53 Patches:
sip_notify_from_rfc3265.patch uploaded by andrew53 (license 519)
* channels/chan_sip.c: Make the sip_standard_port function more
granular by allowing separate type and port arguments. This is
necessary because when building our From and Contact headers, we
need to be absolutely sure that we are placing our source port
there and not the peer's source port. (closes issue #12761)
Reported by: asbestoshead Patches:
patch-chan-sip-contact-port.txt uploaded by asbestoshead (license
455)
* channels/chan_sip.c: Get this compiling in dev-mode
* channels/chan_sip.c: If a peer uses any transport other than UDP,
then MWI will fail for that peer since sip_alloc will allocate a
sip_pvt with a default transport of UDP. This change resets the
socket type immediately after allocating the sip_pvt in
sip_send_mwi_from_peer, so that the proceeding call to
create_addr_from_peer does not fail right away. The socket data
from the peer is properly copied to the sip_pvt in
create_addr_from_peer. (closes issue #13710) Reported by:
andrew53 Patches: sip_notify_use_tcp.patch uploaded by andrew53
(license 519)
* channels/chan_sip.c: When attempting to resolve hostnames, we
need to be sure to remove any parameters from the string so that
name resolution succeeds. (closes issue #13727) Reported by:
fnordian Patches: resolvewithouturiparameter.patch uploaded by
fnordian (license 110)
2008-10-21 15:20 +0000 [r151371] Tilghman Lesher <tlesher@digium.com>
* apps/app_mixmonitor.c: Default file modes should always be full
read and write, to allow the system administrator to make the
decision of what permissions will actually be given, through the
use of the process umask. (Closes issue# 13751)
2008-10-21 11:02 +0000 [r151327] BJ Weschke <bweschke@btwtech.com>
* channels/chan_sip.c: Fix configuration parsing so type=friend
still identifies "friend" as a peer even though it is now a
legacy configuration verb. (closes issue #13705) reported by:
blitzrage patched by: bweschke
2008-10-20 05:07 +0000 [r151246] BJ Weschke <bweschke@btwtech.com>
* pbx/pbx_config.c, main/config.c: Do NOT attempt to do anything
with the ast_config struct when it's been returned as INVALID by
the config file interpreter. (closes issue #13741)
2008-10-20 05:00 +0000 [r151242-151243] Kevin P. Fleming <kpfleming@digium.com>
* autoconf/ast_check_pwlib.m4, /, autoconf/ast_check_openh323.m4,
configure.ac: Merged revisions 151241 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r151241 | kpfleming | 2008-10-20 07:57:33 +0300 (Mon, 20 Oct
2008) | 2 lines rename this macro to properly reflect what it
does ........
* autoconf/ast_prog_egrep.m4, autoconf/ast_c_define_check.m4,
autoconf/ast_ext_tool_check.m4 (added),
autoconf/ast_check_mandatory.m4 (added), /,
autoconf/ast_check_openh323.m4, autoconf/ast_prog_ld_gnu.m4,
autoconf/ast_prog_sed.m4, acinclude.m4 (removed),
autoconf/ast_check_pwlib.m4, autoconf (added),
autoconf/acx_pthread.m4, autoconf/ast_func_fork.m4, configure,
autoconf/ast_gcc_attribute.m4, bootstrap.sh,
autoconf/ast_check_gnu_make.m4, autoconf/ast_ext_lib.m4,
autoconf/ast_prog_ld.m4, autoconf/ast_c_compile_check.m4: Merged
revisions 151240 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r151240 | kpfleming | 2008-10-20 07:45:56 +0300 (Mon, 20 Oct
2008) | 3 lines break up acinclude.m4 into individual files,
which will make it easier to maintain, easier to add new macros
(less patching) and will ease maintenance of these macros across
Asterisk branches ........
2008-10-19 20:30 +0000 [r151188-151190] BJ Weschke <bweschke@btwtech.com>
* /: Block 151167 from coming forward into the /trunk this is a 1.4
fix only.
* /: Block 151100 from coming forward into the /trunk this is a 1.4
fix only.
2008-10-19 19:11 +0000 [r151101] Kevin P. Fleming <kpfleming@digium.com>
* main/tcptls.c, main/manager.c, channels/chan_sip.c, main/http.c,
apps/app_externalivr.c, include/asterisk/tcptls.h: cleaup of the
TCP/TLS socket API: 1) rename 'struct server_args' to 'struct
ast_tcptls_session_args', to follow coding guidelines 2) make
ast_make_file_from_fd() static and rename it to something that
indicates what it really is for (again coding guidelines) 3)
rename address variables inside 'struct ast_tcptls_session_args'
to be more descriptive (dare i say it... coding guidelines) 4)
change ast_tcptls_client_start() to use the new 'remote_address'
field of the session args for the destination of the connection,
and use the 'local_address' field to bind() the socket to the
proper source address, if one is supplied 5) in chan_sip, ensure
that we pass in the PP address we are bound to when creating
outbound (client) connections, so that our connections will
appear from the correct address
2008-10-19 13:10 +0000 [r151060] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_skinny.c: dont segfault when placing a call to a
line that has no registered device.
2008-10-19 07:20 +0000 [r151019] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Adding changes from train and flight back
home from SIPit23 in Lannion, France. - Additional comments on
TCP/TLS implementation - Some additions for new drafts/rfcs (no
new functionality really, mostly documentation) - Other random
small fixes
2008-10-18 10:27 +0000 [r150930-150971] Michiel van Baak <michiel@vanbaak.info>
* Makefile: Make sure we support nested functions and generation of
trampolines under OpenBSD. (closes issue #13724) Reported by:
mvanbaak
* contrib/init.d/rc.mandriva.asterisk,
contrib/init.d/rc.debian.asterisk,
contrib/init.d/rc.redhat.asterisk,
contrib/init.d/rc.suse.asterisk: dont use deprecated commands in
the init scripts. (closes issue #13720) Reported by:
decryptus_proformatique Patches:
contrib_initd_module_reload.patch uploaded by decryptus (license
555) With mods by me to fix stop commands as well
2008-10-18 03:35 +0000 [r150773-150887] BJ Weschke <bweschke@btwtech.com>
* apps/app_authenticate.c, CHANGES: Give app_authenticate the
ability to select a prompt other than the default. (closes issue
#13734) reported and patched by: jvandal
* main/manager.c, /: Using the GetVar handler in AMI is potentially
dangerous (insta-crash [tm]) when you use a dialplan function
that requires a channel and then you don't provide one or provide
an invalid one in the Channel: parameter. We'll handle this
situation exactly the same way it was handled in pbx.c back on
r61766. We'll create a bogus channel for the function call and
destroy it when we're done. If we have trouble allocating the
bogus channel then we're not going to try executing the function
call at all and run the risk of crashing. (closes issue #13715)
reported by: makoto patch by: bweschke
* doc/manager_1_1.txt, CHANGES, apps/app_queue.c: The QueueEntry
event now has the uniqueid of the channel included. (closes issue
#13731) reported and patched by: caio1982
2008-10-17 21:48 +0000 [r150731] Matthew Fredrickson <creslin@digium.com>
* configure, configure.ac: Update configure check to check for new
function in libpri (pri_progress_with_cause)
2008-10-17 21:35 +0000 [r150729] Jason Parker <jparker@digium.com>
* codecs/codec_adpcm.c, codecs/ex_g722.h (added),
codecs/codec_gsm.c, codecs/ex_adpcm.h (added), codecs/ex_alaw.h
(added), codecs/ex_g726.h (added), codecs/ex_gsm.h (added),
codecs/slin_ulaw_ex.h (removed), codecs/slin_lpc10_ex.h
(removed), codecs/codec_resample.c, codecs/slin_g722_ex.h
(removed), codecs/g722_slin_ex.h (removed), codecs/ex_ulaw.h
(added), codecs/adpcm_slin_ex.h (removed), codecs/ex_ilbc.h
(added), codecs/slin_adpcm_ex.h (removed), codecs/g726_slin_ex.h
(removed), codecs/slin_g726_ex.h (removed), codecs/codec_lpc10.c,
codecs/gsm_slin_ex.h (removed), codecs/slin_gsm_ex.h (removed),
codecs/codec_a_mu.c, codecs/codec_g722.c, codecs/ex_lpc10.h
(added), codecs/codec_alaw.c, codecs/codec_speex.c,
codecs/codec_g726.c, include/asterisk/slin.h (added),
codecs/ex_speex.h (added), codecs/slin_resample_ex.h (removed),
codecs/ulaw_slin_ex.h (removed), codecs/slin_ilbc_ex.h (removed),
codecs/ilbc_slin_ex.h (removed), codecs/lpc10_slin_ex.h
(removed), codecs/codec_ulaw.c, codecs/codec_ilbc.c,
codecs/speex_slin_ex.h (removed), codecs/slin_speex_ex.h
(removed): Merge codec_consistency branch. This should make
sample usage much happier.
2008-10-17 17:31 +0000 [r150664] Michiel van Baak <michiel@vanbaak.info>
* main/cli.c: Fix CLI command 'channel request hangup' Prodded on
IRC by Russell and fixed by eliel (closes issue #13730) Reported
by: eliel Patches: main_cli.patch uploaded by eliel (license 64)
2008-10-17 17:25 +0000 [r150640] Matthew Fredrickson <creslin@digium.com>
* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Merge in
patch for #13454. Includes CallRereouting dialplan application,
option for discard of remote hold messages, and using the
alternate logical channel mapping in Q.SIG instead of the default
physical channel mapping.
2008-10-17 17:09 +0000 [r150580-150635] Tilghman Lesher <tlesher@digium.com>
* channels/chan_iax2.c: Make helper call a little safer (suggested
by Russell on IRC)
* include/asterisk/sched.h, channels/chan_iax2.c: Fix the FRACK!
warnings in chan_iax2 when POKE/LAGRQ packets are not answered.
2008-10-17 08:42 +0000 [r150469-150510] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Adding some additional thoughts on
configuration changes to TCP/TLS
* Makefile: Make sure we support nested functions with GCC 4.01
OS/X. This might not be OS/X only, but I'll leave it to kpfleming
to add this to the configure script for testing.
2008-10-17 06:00 +0000 [r150426] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_skinny.c, UPGRADE.txt, configs/skinny.conf.sample,
CHANGES: Break up skinny.conf into seperate sections for devices
and lines. (closes issue #13412) Reported by: wedhorn Patches:
config-restruct-v4.diff uploaded by wedhorn (license 30)
2008-10-17 04:28 +0000 [r150384] Tilghman Lesher <tlesher@digium.com>
* apps/app_meetme.c: Fix option handling code. (closes issue
#11040) Reported by: DEA Patches: rt-meetme-flag-fixes-v2.txt
uploaded by DEA (license 3) with additional fixes by me
2008-10-17 00:18 +0000 [r150311] Mark Michelson <mmichelson@digium.com>
* doc/manager_1_1.txt, CHANGES, channels/chan_iax2.c: Add an
IAXregistry manager command. See doc/manager_1_1.txt for more
details of this command. (closes issue #13326) Reported by: ib2
Patches: bug13326_trunk_20080822.diff uploaded by snuffy (license
35)
2008-10-17 00:14 +0000 [r150309] Jeff Peeler <jpeeler@digium.com>
* apps/app_meetme.c: Initialize character arrays as they are not
guaranteed to be set.
2008-10-17 00:13 +0000 [r150207-150307] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: After a long discussion on #asterisk-bugs,
it seems kind of odd that a channel would be named after the
originating port. For endpoints that always include ":5060" as
part of the From: header, it will mean that you have a ton of
channels with names like "SIP/5060-3ea38a8b." I am boldly moving
forward with this change in trunk, but I'm not touching other
branches with this one since this definitely would qualify as a
behavior change. If there is a problem with this commit, and I
haven't seen the obvious reason why you'd want to name the
channel after the port from which the call originated, then
please feel free to revert this
* main/manager.c, /: Merged revisions 150304 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r150304 | mmichelson | 2008-10-16 18:40:54 -0500 (Thu, 16 Oct
2008) | 6 lines Reverting changes from commits 150298 and 150301
since I was mistakenly under the assumption that dialplan
functions *always* required that a channel be present. I need to
go home earlier, I think :) ........
* main/manager.c: Merged revisions 150298,150301 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r150298 | mmichelson | 2008-10-16 18:34:37 -0500 (Thu, 16 Oct
2008) | 10 lines Don't try to call a dialplan function's read
callback from the manager's GetVar handler if an invalid channel
has been specified. Several dialplan functions, including CHANNEL
and SIP_HEADER, do not check for NULL-ness of the channel being
passed in. (closes issue #13715) Reported by: makoto ........
r150301 | mmichelson | 2008-10-16 18:35:07 -0500 (Thu, 16 Oct
2008) | 3 lines And don't forget to return on the error condition
........
* apps/app_sms.c: Answer the channel prior to checking for the 'a'
option in app_sms. (closes issue #13675) Reported by: alecdavis
Patches: app_sms.bug13675.148985.diff.txt uploaded by alecdavis
(license 585)
* apps/app_skel.c: Updating app_skel.c to follow coding guidelines
with regards to braces used on if statements. (closes issue
#13696) Reported by: alecdavis Patches:
app_skel.bug13696B.115850.diff.txt uploaded by alecdavis (license
585)
* channels/chan_iax2.c: Remove an odd redundant comparison
* configure, configure.ac: Change configure script to search for
openais in both /usr/lib and /usr/lib64 since some distros place
64-bit libraries only in the /usr/lib64 directory. (closes issue
#13721) Reported by: jcollie Patches:
0007-Look-in-64bit-dirs-for-openais.patch uploaded by jcollie
(license 412)
* channels/chan_sip.c: INVITES with proxy auth were sent with a
different branch than what was in the invite_branch of a sip_pvt,
meaning that if a CANCEL were sent later, the branch in the
CANCEL would not match the branch in the latest INVITE sent out,
leading to some endpoints responding to the CANCEL with a 481.
(closes issue #13714) Reported by: fnordian Patches:
invite_branch.patch uploaded by fnordian (license 110)
2008-10-16 16:04 +0000 [r150125] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c, /: Merged revisions 150124 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r150124 | rmudgett | 2008-10-16 10:56:06 -0500 (Thu, 16
Oct 2008) | 1 line Fix memory leak found by customer ........
2008-10-16 15:48 +0000 [r150118-150121] Terry Wilson <twilson@digium.com>
* configs/modules.conf.sample: This is nolonger needed
* res/res_phoneprov.c: func_strings isn't a dependency of this
module anymore
2008-10-16 15:02 +0000 [r150052] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_sip.c: ensure that type=peer entries are only
matched on IP/port, not on name (after oej audits all the calls
to find_peer() to make sure that forcenamematch is set correctly
in each case)
2008-10-16 15:00 +0000 [r150008-150051] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Doxygen addition
* channels/chan_sip.c: Add some notes on problems with the TCP/TLS
implementation
2008-10-16 13:28 +0000 [r149917-149981] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_sip.c: return this logic to where it used to be,
*after* the dialog->needdestroy flag has been determined to be
set; otherwise, we generate these debug messages every time we
inspect every active dialog
* channels/chan_sip.c: some additional debugging tools added at
SIPit23: - move all setting of 'needdestroy' on dialog structures
into the history - report all tags involved when a pedantic check
fails on a REFER
* res/res_phoneprov.c: inter-module dependencies should be included
in the source code, not just in sample config files
* res/res_phoneprov.c: correct file name in message
* configs/musiconhold.conf.sample, res/res_musiconhold.c, CHANGES:
support relative paths in musiconhold.conf, which makes moh work
by default when Asterisk was configured using --prefix and 'make
samples' is run
2008-10-15 21:36 +0000 [r149848] BJ Weschke <bweschke@btwtech.com>
* /: Blocking 149840 from coming forward.
2008-10-15 20:55 +0000 [r149802] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Make the sip_proxy struct reference counted.
This is necessary to allow for a sip_pvt to maintain a reference
to a sip_peer's outboundproxy even after the peer has been freed.
(closes issue #13700) Reported by: fnordian Patches: 13700.patch
uploaded by putnopvut (license 60) Tested by: fnordian
2008-10-15 20:14 +0000 [r149756] BJ Weschke <bweschke@btwtech.com>
* configs/agents.conf.sample, /: Merged revisions 149683 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r149683 | bweschke | 2008-10-15 14:28:54 -0400 (Wed, 15 Oct 2008)
| 4 lines An update to the documentation/example of
agents.conf.sample with the correct parameter for this feature as
defined in chan_agent.c (closes issue #13709) ........
2008-10-15 19:07 +0000 [r149588-149687] Tilghman Lesher <tlesher@digium.com>
* funcs/func_odbc.c: Permit data fields to contain more than 255
characters. (closes issue #13631) Reported by: seanbright
Patches: 20081015__bug13631.diff.txt uploaded by Corydon76
(license 14) Tested by: blitzrage
* funcs/func_odbc.c: Only set buf to blank before the goto.
* codecs/lpc10/lpcini.c: When using MALLOC_DEBUG, codec_lpc10 leaks
memory, because it matches a library malloc() with an ast_free
(which, of course, doesn't match up with known allocated memory,
so the free fails). (closes issue #13702) Reported by: eliel
Patches: codec_lpc10_lpcini.c uploaded by eliel (license 64)
* apps/app_echo.c: Minor spacing change (closes issue #13697)
Reported by: alecdavis Patches: app_echo.bug13697.103249.diff.txt
uploaded by alecdavis (license 585)
2008-10-15 13:52 +0000 [r149542] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Adding a note about a missing part of
"kill-the-user" - I got lost in the Ao2 world... We're going to
try to get time to fix this and kpfleming believes that there's
code in ao2 so that we can solve it...
2008-10-15 11:26 +0000 [r149384-149487] Kevin P. Fleming <kpfleming@digium.com>
* /, channels/chan_sip.c: Merged revisions 149452 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r149452 | kpfleming | 2008-10-15 12:30:40 +0200 (Wed, 15 Oct
2008) | 3 lines fix some problems when parsing SIP messages that
have the maximum number of headers or body lines that we support
........
* configure, configure.ac: reverting this change... had not read
the commit list yet, didn't realize the code had been upgraded
* configure, configure.ac: do complete version check for SpanDSP,
since the app_fax code is not compatible with 0.0.6 yet
* apps/app_stack.c: building this module depends on res_agi being
built as well
2008-10-15 07:45 +0000 [r149342] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Fixing sytax errors ;-)
2008-10-14 23:57 +0000 [r149201-149279] Mark Michelson <mmichelson@digium.com>
* apps/app_dial.c, CHANGES: When specifying an invalid timeout to
Dial, take it to mean that no timeout is desired. (closes issue
#13625) Reported by: atis
* /, channels/chan_sip.c: Merged revisions 149266 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r149266 | mmichelson | 2008-10-14 18:43:58 -0500 (Tue, 14 Oct
2008) | 4 lines Change this warning to an error message.
Suggestion comes from Sean Bright. Thanks Sean! ........
* /, channels/chan_sip.c: Merged revisions 149207 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r149207 | mmichelson | 2008-10-14 18:10:26 -0500 (Tue, 14 Oct
2008) | 9 lines Call register_peer_exten even in the case that
the peer's IP/port does not change. (closes issue #13309)
Reported by: dimas Patches: v2-13309.patch uploaded by dimas
(license 88) ........
* /, include/asterisk/audiohook.h, main/audiohook.c: Merged
revisions 149204 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r149204 | mmichelson | 2008-10-14 18:00:01 -0500 (Tue, 14 Oct
2008) | 12 lines Add a tolerance period for sync-triggered
audiohooks so that if packetization of audio is close (but not
equal) we don't end up flushing the audiohooks over small
inconsistencies in synchronization. Related to issue #13005, and
solves the issue for most people who were experiencing the
problem. However, a small number of people are still experiencing
the problem on long calls, so I am not closing the issue yet
........
* /, apps/app_queue.c: Merged revisions 149200 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r149200 | mmichelson | 2008-10-14 17:40:42 -0500 (Tue, 14 Oct
2008) | 12 lines Update the queue with the correct number of
calls and whether the call was completed within the service level
when a transfer takes place. This way, we do not "break" the
leastrecent and fewestcalls strategies by not logging a call
until after the transferred call has ended. (closes issue #13395)
Reported by: Marquis Patches: app_queue.c.transfer.patch uploaded
by Marquis (license 32) ........
2008-10-14 22:38 +0000 [r149199] Tilghman Lesher <tlesher@digium.com>
* main/hashtab.c, pbx/pbx_spool.c, channels/chan_sip.c,
include/asterisk/chanvars.h, include/asterisk/config.h,
include/asterisk/strings.h, res/res_indications.c,
include/asterisk/hashtab.h, main/chanvars.c, main/config.c: Add
additional memory debugging to several core APIs, and fix several
memory leaks found with these changes. (Closes issue #13505,
closes issue #13543) Reported by: mav3rick, triccyx Patches:
20081001__bug13505.diff.txt uploaded by Corydon76 (license 14)
Tested by: mav3rick, triccyx
2008-10-14 21:08 +0000 [r149131] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 149130 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r149130 | mmichelson | 2008-10-14 15:49:02 -0500 (Tue, 14 Oct
2008) | 7 lines Don't allow reserved characters to be used in
register lines in sip.conf. (closes issue #13570) Reported by:
putnopvut ........
2008-10-14 20:16 +0000 [r149062] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_waitforsilence.c: Merged revisions 149061 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r149061 | tilghman | 2008-10-14 15:09:06 -0500 (Tue, 14 Oct 2008)
| 6 lines Check correct values in the return of ast_waitfor();
also, get rid of a possible memory leak. (closes issue #13658)
Reported by: explidous Patch by: me ........
2008-10-14 19:35 +0000 [r149040] Leif Madsen <lmadsen@digium.com>
* doc/manager_1_1.txt: Add missing documentation for
SipShowRegistry action and RegistryEntry event. (closes issue
#13342) Reported and patch by: Laureano
2008-10-14 19:03 +0000 [r148917-148988] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_voicemail.c: Merged revisions 148987 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r148987 | tilghman | 2008-10-14 14:03:08 -0500 (Tue, 14
Oct 2008) | 2 lines Some compilers warn, some don't. Fixing.
........
* apps/app_sms.c: App is ignoring 'p' parameter -- initial pause.
(closes issue #13617) Reported by: alecdavis Patches:
app_sms.13oct.diff.txt uploaded by alecdavis (license 585)
* /, apps/app_voicemail.c: Merged revisions 148916 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r148916 | tilghman | 2008-10-14 12:41:08 -0500 (Tue, 14
Oct 2008) | 4 lines Ensure that mail headers are 7-bit clean,
even when UTF-8 characters are used in headers like 'Subject' and
'To'. Closes AST-107. ........
2008-10-14 17:38 +0000 [r148913] Mark Michelson <mmichelson@digium.com>
* channels/chan_local.c, /: Merged revisions 148912 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r148912 | mmichelson | 2008-10-14 12:33:38 -0500 (Tue,
14 Oct 2008) | 9 lines Deadlock prevention in chan_local. (closes
issue #13676) Reported by: tacvbo Patches: 13676.patch uploaded
by putnopvut (license 60) Tested by: tacvbo ........
2008-10-14 15:15 +0000 [r148868] Tilghman Lesher <tlesher@digium.com>
* apps/app_fax.c: API differences in spandsp 0.0.6pre1 and higher
(closes issue #13688) Reported by: irroot Patches:
app_fax-span6.patch uploaded by irroot (license 52) with minor
modifications by me
2008-10-14 15:00 +0000 [r148867] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Fix reference count issue that Russell
brought up in SIP MWI NOTIFY support. Bump the reference count up
before we add it to the scheduler, duh.
2008-10-14 14:18 +0000 [r148825] Doug Bailey <dbailey@digium.com>
* phoneprov/polycom.xml: Allow MWI registration for all configured
lines.
2008-10-14 11:31 +0000 [r148695-148754] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_sip.c: fix some references to the owner of a
private structure that may not be present
* Makefile, /: Merged revisions 148736 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r148736 | kpfleming | 2008-10-14 12:30:54 +0200 (Tue, 14 Oct
2008) | 3 lines on Ubuntu (at least), recent versions of ld in
binutils delete all debugging symbols when -x is supplied; since
the reasons why -x is being passed are lost in the mists of time,
remove it so debugging will work properly ........
* channels/chan_sip.c: this structure should be static
* channels/chan_sip.c: ensure that *all* fields in the req
structure are cleared out before reusing it; has_to_tag was not
cleared, which caused the second incoming call over a TCP socket
to fail if pedantic checking was enabled
2008-10-14 09:16 +0000 [r148679] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Adding some clarifications
2008-10-14 08:06 +0000 [r148612] Kevin P. Fleming <kpfleming@digium.com>
* /, main/translate.c: Merged revisions 148611 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r148611 | kpfleming | 2008-10-14 02:54:41 -0500 (Tue, 14 Oct
2008) | 3 lines it would be nice if this message printing code
had actually been tested before it was committed... ........
2008-10-14 00:08 +0000 [r148570] Tilghman Lesher <tlesher@digium.com>
* res/res_config_curl.c, res/res_config_pgsql.c,
res/res_config_odbc.c, include/asterisk/config.h,
res/res_realtime.c, include/asterisk/strings.h,
res/res_config_ldap.c, res/res_config_sqlite.c, main/config.c,
apps/app_voicemail.c: Merge realtime_update2 branch, which adds a
new realtime API call named 'update2', which permits updates
which match across multiple columns, instead of requiring all
tables to have a single unique identifier. All of the other API
calls with the exception of 'update' already had the ability to
match on multiple fields, so it was a missing and very desireable
feature that an API call implementing an update should have this,
too. This does not change any outward performance of Asterisk,
but it should make life easier for application developers who use
the RealTime framework.
2008-10-13 17:14 +0000 [r148519] Steve Murphy <murf@digium.com>
* main/pbx.c: Hmmm. Nobody (but me) is interested in seeing the
trie info when they do 'dialplan show ...' (even with debug set
to non-zero); so I set up a 'dialplan debug [context]' cli
command instead, to explicitly show just the trie info. I even
added an extension_exists() call to make sure the trie info is
built. I moved the explanatory header to above the extension loop
to ensure it only prints once. And it will do this now, whether
debug is set or not. I removed the trie printing from the
'dialplan show' command entirely.
2008-10-13 15:56 +0000 [r148471-148474] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: - Doxygen formatting. (tss tss) - Fixing
language
* main/tcptls.c, channels/chan_sip.c: Highlightning even more bugs
in the current tcp/tls implementation.
* channels/chan_sip.c: Sending a 403 after a 200 is considered very
bad. (found at SIPit)
2008-10-12 09:19 +0000 [r148425] Michiel van Baak <michiel@vanbaak.info>
* res/res_agi.c: fix the 'agi show commands' CLI function. (closes
issue #13666) Reported by: eliel Patches: res_agi.c.patch
uploaded by eliel (license 64)
2008-10-10 21:21 +0000 [r148373-148376] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: The logic used when checking a peer got
changed subtly in the "kill the user" commit and caused calls
relying on the insecure setting to not work properly. I changed
for finding a peer back to how it was prior to that commit.
(closes issue #13644) Reported by: pj Patches:
13644_trunkv2.patch uploaded by putnopvut (license 60) Tested by:
pj
* channels/chan_sip.c: Make sure that the inUse and inRinging
fields for a sip peer cannot go below zero. This is a regression
from 1.4 and so it will be applied to 1.6.0 as well. (closes
issue #13668) Reported by: mjc
2008-10-10 18:59 +0000 [r148268-148329] Tilghman Lesher <tlesher@digium.com>
* pbx/pbx_config.c: Reset continuation items at the beginning of
each context (suggested by kpfleming).
* CHANGES, pbx/pbx_config.c: Add keyword "same", which allows you
to create multiple steps in a dialplan, without needing to
respecify an extension pattern multiple times. (closes issue
#13632) Reported by: blitzrage Patches:
20081006__bug13632.diff.txt uploaded by Corydon76 (license 14)
Tested by: blitzrage, Corydon76
* /, apps/app_voicemail.c: Merged revisions 148257 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r148257 | tilghman | 2008-10-10 11:25:31 -0500 (Fri, 10
Oct 2008) | 7 lines User not notified of temporary greeting, if
ODBC storage is in use. (closes issue #13659) Reported by:
moliveras Patches: 20081009__bug13659.diff.txt uploaded by
Corydon76 (license 14) Tested by: moliveras ........
2008-10-10 00:42 +0000 [r148200] Sean Bright <sean.bright@gmail.com>
* include/asterisk.h, main/tdd.c, main/cryptostub.c,
res/res_config_sqlite.c, apps/app_voicemail.c: Don't include
logger.h in asterisk.h by default as it is causing problems
building app_voicemail. Instead, include it where it is needed.
This turned out to be a relatively minor issue because other
headers include logger.h as well. Need to test -addons before
merging this back to 1.6.0. (closes issue #13605) Reported by:
tomo1657 Patches: 13605_seanbright.diff uploaded by seanbright
(license 71) Tested by: mmichelson
2008-10-09 23:54 +0000 [r148144-148160] Mark Michelson <mmichelson@digium.com>
* main/manager.c: The priority was unnecessary for the manager
atxfer, so it has been removed. Furthermore, now we actually use
the Context argument passed to set the transfer context and don't
error out if no context is specified. This addresses the actual
problems outlined in issue 12158. Regarding the other points
brought up, regarding the inability to not transfer to extensions
which cannot be represented by DTMF, it is not enough of a
constraint that it is worth attempting to rework the feature.
(closes issue #12158) Reported by: davidw
* apps/app_voicemail.c: Read the callerid in the correct order and
make sure to read the Urgent flag value from the IMAP headers.
(closes issue #13652) Reported by: jaroth Patches:
imapheaders.patch uploaded by jaroth (license 50)
2008-10-09 23:25 +0000 [r148120] Tilghman Lesher <tlesher@digium.com>
* configs/res_ldap.conf.sample: Fix example schema (closes issue
#12860) Reported by: flyn Patches: res_ldap.conf.patch uploaded
by flyn (license 503)
2008-10-09 23:15 +0000 [r148112] Mark Michelson <mmichelson@digium.com>
* /, main/features.c: Merged revisions 146026 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r146026 | murf | 2008-10-03 12:12:54 -0500 (Fri, 03 Oct 2008) |
18 lines (closes issue #13579) Reported by: dwagner (closes issue
#13584) Reported by: dwagner Tested by: murf, putnopvut The
thought occurred to me that the res= from the extension spawn was
ending up being returned from the bridge. "Thou shalt not poison
the return value". Made the change and it appears to allow blind
xfers to work as normal. If I'm wrong, reopen the bugs. But it
looks good to me! Many thanks to putnopvut for helping me
reproduce this! ........
2008-10-09 21:47 +0000 [r148000-148071] Tilghman Lesher <tlesher@digium.com>
* formats/format_wav.c, apps/app_minivm.c, channels/chan_agent.c,
main/file.c, res/res_monitor.c, apps/app_voicemail.c: Reverting
format addition for now
* apps/app_minivm.c, channels/chan_agent.c, main/file.c,
res/res_monitor.c, apps/app_voicemail.c: Fudges for wav16, just
like wav49
* formats/format_wav.c: Add native 16kHz format for wav file
format. (Closes issue #13657)
* sounds/sounds.xml, sounds/Makefile: Publish MOH files in sln16
format
* /, apps/app_voicemail.c: Merged revisions 147997 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r147997 | tilghman | 2008-10-09 14:38:33 -0500 (Thu, 09
Oct 2008) | 4 lines When blank, callerid name and number should
display "unknown caller" in voicemail emails. (Closes issue
#13643) ........
2008-10-09 19:27 +0000 [r147952] Jeff Peeler <jpeeler@digium.com>
* main/features.c: (closes issue #13139) Reported by: krisk84
Tested by: krisk84 This change prevents a call that is placed in
the parkinglot to be picked up before the PBX is finished. If
another extension dials the parking extension before the PBX
thread has completed at minimum warnings will occur about the PBX
not properly being terminated. At worst, a crash could occur.
2008-10-09 17:48 +0000 [r147899] Michiel van Baak <michiel@vanbaak.info>
* include/asterisk/endian.h: only include this for OpenBSD. At
least FreeBSD is borked when including it (closes issue #13649)
Reported by: ys
2008-10-09 17:46 +0000 [r147896] Tilghman Lesher <tlesher@digium.com>
* configs/extensions.conf.sample: Remove "second form" of
extensions, as it no longer applies. Also, cleanup the grammar,
formatting, and introduce several clarifications to the text.
(Closes issue #13654)
2008-10-09 17:04 +0000 [r147854] Terry Wilson <twilson@digium.com>
* phoneprov/000000000000.cfg, res/res_phoneprov.c,
configs/phoneprov.conf.sample: Make phoneprov case-insensitive to
remove the func_strings dependency of the default config
2008-10-09 17:01 +0000 [r147853] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_dahdi.c, channels/chan_misdn.c,
channels/chan_h323.c: fix some CLI commands we borked during
devcon2008 Thanks rmudget for letting me know and providing hints
on how to fix it best.
2008-10-09 14:17 +0000 [r147807] Steve Murphy <murf@digium.com>
* main/pbx.c, include/asterisk.h, doc/CODING-GUIDELINES,
include/asterisk/autoconfig.h.in, channels/vcodecs.c,
main/translate.c, configure.ac, channels/console_video.c,
channels/chan_iax2.c, main/astobj2.c, channels/chan_oss.c,
main/rtp.c, main/config.c, main/cli.c, channels/chan_usbradio.c,
configure, channels/console_gui.c, utils/extconf.c: (closes issue
#13557) Reported by: nickpeirson Patches: pbx.c.patch uploaded by
nickpeirson (license 579) replace_bzero+bcopy.patch uploaded by
nickpeirson (license 579) Tested by: nickpeirson, murf 1.
replaced all refs to bzero and bcopy to memset and memmove
instead. 2. added a note to the CODING-GUIDELINES 3. add two
macros to asterisk.h to prevent bzero, bcopy from creeping back
into the source 4. removed bzero from configure, configure.ac,
autoconfig.h.in
2008-10-09 01:43 +0000 [r147760-147761] Joshua Colp <jcolp@digium.com>
* configs/sip.conf.sample: *whistle*
* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add
support for subscribing to a voice mailbox on a remote SIP server
and making the new/old message count available to local devices.
(issue #AST-77)
2008-10-08 22:32 +0000 [r147714] Mark Michelson <mmichelson@digium.com>
* apps/app_meetme.c: Some small tweaks regarding realtime
conference announcements. (closes issue #13522) Reported by: DEA
Patches: meetme-rt-fixes.txt uploaded by DEA (license 3)
2008-10-08 22:26 +0000 [r147689] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 147681 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r147681 | kpfleming | 2008-10-08 17:22:09 -0500 (Wed, 08
Oct 2008) | 3 lines when parsing a text configuration option,
ensure that the buffer on the stack is actually large enough to
hold the legal values of that option, and also ensure that
sscanf() knows to stop parsing if it would overrun the buffer
(without these changes, specifying "buffers=...,immediate" would
overflow the buffer on the stack, and could not have worked as
expected) ........
2008-10-08 20:07 +0000 [r147635] Sean Bright <sean.bright@gmail.com>
* configs/voicemail.conf.sample: Add some examples of IMAP
accounts.
2008-10-08 19:08 +0000 [r147592] Tilghman Lesher <tlesher@digium.com>
* apps/app_sms.c: Correct a typo in the help; also, ensure that the
date and time are correctly set, if not specified in the message.
(Closes issue #13594, closes issue #13595) Reported by: alecdavis
Patches: 20081001__bug13595.diff.txt uploaded by Corydon76
(license 14) Tested by: alecdavis
2008-10-08 14:53 +0000 [r147518] Joshua Colp <jcolp@digium.com>
* /, apps/app_speech_utils.c: Merged revisions 147517 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r147517 | file | 2008-10-08 11:51:42 -0300 (Wed, 08 Oct
2008) | 2 lines If we receive DTMF make sure that the state of
the speech structure goes back to being not ready. (issue
#LUMENVOX-8) ........
2008-10-08 12:28 +0000 [r147476] Bradley Latus <brad.latus@gmail.com>
* configs/iax.conf.sample: Adjust commented default trunkmtu value
to match documentation above it
2008-10-08 12:15 +0000 [r147388-147457] Sean Bright <sean.bright@gmail.com>
* funcs/func_curl.c, apps/app_meetme.c, cdr/cdr_adaptive_odbc.c,
res/res_odbc.c: Keep up with shadow warnings. One day I'll
actually enable this in the Makefile.
* utils/Makefile: When echoing our copies, strip off ASTTOPDIR from
the front of the source file.
* apps/app_dial.c, channels/chan_dahdi.c, channels/chan_iax2.c:
Move the DAHDI-to-DAHDI operator mode check from app_dial into
chan_dahdi so we don't have to hardcode anything. (closes issue
#13636) Reported by: seanbright Patches: 13636.diff uploaded by
seanbright (license 71) Reviewed by: russellb, putnopvut
2008-10-07 20:15 +0000 [r147266-147347] Michiel van Baak <michiel@vanbaak.info>
* configure, configure.ac: Make format_vorbis_ogg compile on
OpenBSD (closes issue #13639) Reported by: mvanbaak Patches:
2008100700_oggsupportOBSD.diff.txt uploaded by mvanbaak (license
7) 2008100700_oggsupportOBSD-configurescript.diff.txt uploaded by
mvanbaak (license 7) Tested by: mvanbaak
* channels/Makefile: make this work on OpenBSD
* configure, configure.ac: Make sure the configs on OpenBSD are in
/etc/asterisk by default (closes issue #13641) Reported by: jtodd
* contrib/scripts/safe_asterisk_restart,
contrib/scripts/safe_asterisk: use pkill instead of killall to be
more portable
2008-10-07 18:00 +0000 [r147265] Sean Bright <sean.bright@gmail.com>
* apps/app_voicemail.c: This was flawed. The issue that I was
trying to address was addressed by adding the imapsecret alias
for imappassword. Will rethink this one and give it another shot
on a rainy day TBD.
2008-10-07 17:49 +0000 [r147264] Michiel van Baak <michiel@vanbaak.info>
* CHANGES: fix wording as pointed out by Corydon
2008-10-07 17:44 +0000 [r147262] Tilghman Lesher <tlesher@digium.com>
* UPGRADE.txt, include/asterisk/options.h, main/asterisk.c,
main/term.c: Allow people to select the old console behavior of
white text on a black background, by using the startup flag '-B'.
2008-10-07 16:52 +0000 [r147191-147194] Sean Bright <sean.bright@gmail.com>
* /, apps/app_voicemail.c: Merged revisions 147193 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r147193 | seanbright | 2008-10-07 12:48:30 -0400 (Tue,
07 Oct 2008) | 2 lines Make 'imapsecret' an alias to
'imappassword' in voicemail.conf. ........
* apps/app_voicemail.c: Or not.
* apps/app_voicemail.c: There was a boo-boo in TFOT that is causing
some confusion on the mailing lists so include 'imapsecret' as an
alias to 'imappassword' (and print a little notice nudging users
toward the right option name).
2008-10-07 16:04 +0000 [r147146] Jeff Peeler <jpeeler@digium.com>
* main/features.c: Explicitly setting these fields to NULL was done
because I wasn't sure if they would be NULL otherwise. Since they
will be set automatically, removing.
2008-10-07 14:59 +0000 [r147050-147099] Sean Bright <sean.bright@gmail.com>
* apps/app_voicemail.c: If we encounter something in mailbox
options that we don't grok, then spit out a warning instead of
just silently ignoring it.
* apps/app_dial.c: Make sure to compare the correct number of
characters when special-casing our DAHDI operator mode stuff.
Technically, it would work fine, as 'DAH' is currently unique
amongst our channel technologies, but as Jared points out:
<@jsmith> Sure... as long as the technology starts whith DAH....
but it could be DAHDOO!
2008-10-07 02:02 +0000 [r147011] Richard Mudgett <rmudgett@digium.com>
* funcs/func_callerid.c: Independent change from branch issue8824
that is not part of COLP. (-r142574 rmudgett)
2008-10-07 00:02 +0000 [r146970] Terry Wilson <twilson@digium.com>
* channels/chan_sip.c: A blind transfer to the parking thread would
cause a segfault because copy_request accesses dst->data w/o
being able to tell whether it is proerly initialized
2008-10-06 23:21 +0000 [r146928] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/threadstorage.h: Update documentation;
AST_THREADSTORAGE() in trunk only takes a single argument.
2008-10-06 23:14 +0000 [r146925] Michiel van Baak <michiel@vanbaak.info>
* res/res_config_odbc.c, build_tools/menuselect-deps.in, configure,
funcs/func_odbc.c, include/asterisk/autoconfig.h.in,
configure.ac, cdr/cdr_adaptive_odbc.c, cdr/cdr_odbc.c,
makeopts.in, res/res_odbc.c, apps/app_voicemail.c: All ODBC parts
can now use either unixodbc or iodbc. This allows for the ODBC
parts to work on OpenBSD as well. 99.99% of the work is done by
seanbright (bow, bow) and I actually did nothing but test and
yell at him that it still didn't work :) Thanks for helping out !
2008-10-06 23:08 +0000 [r146875-146923] Jeff Peeler <jpeeler@digium.com>
* main/features.c, res/res_agi.c, include/asterisk/features.h:
Similar to r143204, masquerade the channel in the case of Park
being called from AGI.
* include/asterisk/endian.h: Mvanbaak said this was needed to
compile on OpenBSD, so put it in the OpenBSD section.
* main/features.c: This commit squashes together three commits
because the wrong approach was originally used. (One of the
commits was only one line.) 1) r143204: The main change here was
to masquerade the channel if the channel that was to be parked
was running a PBX on it. The PBX thread can then maintain full
control of the channel (the zombie) as it expects to while
allowing the parking thread full control of the real (parked)
channel. 2) r143270: Changed park_call_full to hold the
parkinglot lock a little longer, which protects the parkeduser
struct from being freed out from underneath. Made sure that the
parking extension is added to the parking context while holding
the lock thereby ensuring that there are no spurious warnings
from removal attempts when a hangup occurs while the parking lot
is being announced. 3) r143475: (the one liner) compare peer and
chan instead of looking at the parked user (pu), which could have
possibly already have been freed by the parking thread
* main/features.c: fix some comment placement
* main/features.c: Explicitly set args in park_call_exec NULL so in
the case of no options being passed in, there is no garbage
attempted to be used. Also, do not set args to unknown value
again if there are no options passed in.
2008-10-06 21:18 +0000 [r146807] Michiel van Baak <michiel@vanbaak.info>
* include/asterisk/endian.h: make aescrypt.c compile on OpenBSD
again
2008-10-06 21:09 +0000 [r146802] Tilghman Lesher <tlesher@digium.com>
* funcs/func_curl.c, funcs/func_groupcount.c, res/res_smdi.c, /,
channels/chan_sip.c, funcs/func_timeout.c, funcs/func_odbc.c,
funcs/func_cdr.c, funcs/func_math.c, channels/chan_iax2.c,
funcs/func_callerid.c, apps/app_speech_utils.c: Merged revisions
146799 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r146799 | tilghman | 2008-10-06 15:52:04 -0500 (Mon, 06 Oct 2008)
| 8 lines Dialplan functions should not actually return 0, unless
they have modified the workspace. To signal an error (and no
change to the workspace), -1 should be returned instead. (closes
issue #13340) Reported by: kryptolus Patches:
20080827__bug13340__2.diff.txt uploaded by Corydon76 (license 14)
........
2008-10-06 17:32 +0000 [r146738] Sean Bright <sean.bright@gmail.com>
* configure, configure.ac: Pretty-print a couple configure options
2008-10-06 16:52 +0000 [r146713] Tilghman Lesher <tlesher@digium.com>
* channels/chan_local.c, /: Merged revisions 146711 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r146711 | tilghman | 2008-10-06 11:51:21 -0500 (Mon, 06
Oct 2008) | 9 lines Check whether an extension exists in the
_call method, rather than the _alloc method, because we need to
evaluate the callerid (since that data affects whether an
extension exists). (closes issue #13343) Reported by: efutch
Patches: 20080915__bug13343.diff.txt uploaded by Corydon76
(license 14) Tested by: efutch ........
2008-10-06 16:03 +0000 [r146644] Kevin P. Fleming <kpfleming@digium.com>
* /: Merged revisions 146643 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r146643 | kpfleming | 2008-10-06 10:57:49 -0500 (Mon, 06 Oct
2008) | 8 lines ensure that the private structure for pseudo
channels is created without 'leaking' configuration data from
other configured channels (closes issue #13555) Reported by:
jeffg Patches: issue_13555.patch uploaded by kpfleming (license
421) Tested by: jeffg ........
2008-10-06 15:29 +0000 [r146640] Mark Michelson <mmichelson@digium.com>
* configs/queues.conf.sample, CHANGES, apps/app_queue.c: This
commit introduces a change to how the "joinempty" and
"leavewhenempty" options are configured in queues.conf. Instead
of using vague terms like "yes," "no," "loose," and "strict," we
now accept a comma-separated list of values to determine when to
consider a member available. Extended details can be found in the
queues.conf.sample file. Note also that the above four referenced
values are still accepted for backwards-compatibility, but are
mapped internally to the new method of representing the option.
AST-105
2008-10-06 00:36 +0000 [r146555-146597] Sean Bright <sean.bright@gmail.com>
* utils/Makefile: Make NOISY_BUILD work for the calls to cp in
utils/Makefile
* utils/Makefile: Quote arguments to cp so we can handle spaces in
our paths.
2008-10-05 22:11 +0000 [r146514] Russell Bryant <russell@digium.com>
* utils/muted.c: Make this build on my mac.
2008-10-05 21:21 +0000 [r146449] Jason Parker <jparker@digium.com>
* /, channels/chan_sip.c: Recorded merge of revisions 146448 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r146448 | qwell | 2008-10-05 16:17:44 -0500 (Sun, 05 Oct 2008) |
1 line Fix silly formatting. ........
2008-10-05 01:59 +0000 [r146312-146407] Sean Bright <sean.bright@gmail.com>
* build_tools/make_buildopts_h: This is far from optimal, but I
just found a FreeBSD system without md5 installed on it. So look
around for all of the different binaries that we could possibly
use. I'd wager this gets completely replaced by someone else in
less than 24 hours... :)
* main/asterisk.c: Fix a bug with the last item in CLI history
getting duplicated when read from the .asterisk_history file (and
subsequently being duplicated when written). We weren't checking
the result of fgets() which meant that we read the same line
twice before feof() actually returned non- zero. Also, stop
writing out an extra blank line between each item in the history
file, fix a minor off-by-one error, and use symbolic constants
rather than a hardcoded integer.
* configs/sip_notify.conf.sample: Add ability to remotely reboot
snom phones. Also cleaned up and reorganized
sip_notify.conf.sample a bit as well. Tested snom reboot on snom
360 and verified snom-check-cfg worked as well. (closes issue
#13601) Reported by: mjc Tested by: seanbright
2008-10-03 22:40 +0000 [r146242] Jeff Peeler <jpeeler@digium.com>
* main/features.c: remove superfluous reference counting operations
in manage_parkinglot since ao2_interator_next increments the ref
count automatically
2008-10-03 22:10 +0000 [r146198] Sean Bright <sean.bright@gmail.com>
* main/cli.c: Resolve a subtle bug where we would never
successfully be able to get the first item in the CLI entry list.
This was preventing '!' from showing up in either 'help' or in
tab completion. (closes issue #13578) Reported by: mvanbaak
2008-10-03 18:30 +0000 [r146081] Tilghman Lesher <tlesher@digium.com>
* CHANGES: document meetme schedule changes (related to issue
#11040)
2008-10-03 17:36 +0000 [r146053] Michiel van Baak <michiel@vanbaak.info>
* CHANGES: put a note in CHANGES about the cli_cleanup done during
AstriDevCon
2008-10-03 17:35 +0000 [r146052] Terry Wilson <twilson@digium.com>
* main/dial.c: The dialing API should inherit datastores as well as
variables
2008-10-02 19:30 +0000 [r145959-145962] Russell Bryant <russell@digium.com>
* CHANGES: The 'P' command for ExternalIVR was also added in 1.6.0
* CHANGES: TCP support for ExternalIVR went in to 1.6.1, not 1.6.0
2008-10-02 18:02 +0000 [r145915] Michiel van Baak <michiel@vanbaak.info>
* apps/app_meetme.c: fix the 'meetme list', 'meetme list concise',
'meetme list $confno' and 'meetme list $confno concise' CLI
commands (closes issue #13586) Reported by: john8675309 Help and
feedback from eliel, thanks!
2008-10-02 17:16 +0000 [r145846] Tilghman Lesher <tlesher@digium.com>
* configs/func_odbc.conf.sample, funcs/func_odbc.c, CHANGES: Permit
the syntax and synopsis fields to be set (for func_odbc).
2008-10-02 16:42 +0000 [r145842] Michiel van Baak <michiel@vanbaak.info>
* apps/app_meetme.c: make this compile under devmode again
2008-10-02 15:28 +0000 [r145771] Sean Bright <sean.bright@gmail.com>
* configure, configure.ac: This is much cleaner, methinks.
2008-10-02 15:17 +0000 [r145752] Tilghman Lesher <tlesher@digium.com>
* /, res/res_odbc.c: Merged revisions 145751 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r145751 | tilghman | 2008-10-02 10:13:21 -0500 (Thu, 02 Oct 2008)
| 3 lines Some sanity checks that may have led to prior crashes,
found by codefreeze-lap (murf) on IRC. Also some cleanup of
incorrectly-used constants. ........
2008-10-01 23:48 +0000 [r145692] Sean Bright <sean.bright@gmail.com>
* configure, configure.ac: Try a test compile using the GMime
library. Some distros install gmime-config in the base package
instead of the -devel package. Now we print a notice and disable
GMime support instead of bombing during the main compilation.
(closes issue #13583) Reported by: arkadia
2008-10-01 23:02 +0000 [r145649] Tilghman Lesher <tlesher@digium.com>
* apps/app_meetme.c, funcs/func_strings.c,
include/asterisk/localtime.h, main/stdtime/localtime.c: Add
schedule extensions to app_meetme. In addition, the reporter
found a problem within strptime(3), which we are correcting here
with ast_strptime(). (closes issue #11040) Reported by: DEA
Patches: 20080910__bug11040.diff.txt uploaded by Corydon76
(license 14) Tested by: DEA
2008-10-01 22:23 +0000 [r145553-145606] Mark Michelson <mmichelson@digium.com>
* main/features.c: Okay, this should really do it now. While I did
manage to fix blind transfers with my last commit here, I also
caused an unwanted side-effect. That is, only the first priority
of the 'h' extension would be executed when a blind transfer
occurred instead of all priorities. Essentially, my last commit
corrected the return value of ast_bridge_call. However, the
implementation still was not 100% correct. Now it is.
* main/features.c: if (!(x) == 0) is the same as if (x).
* main/features.c: The logic surrounding the return value of
ast_spawn_extension within ast_bridge_call was reversed. This
problem was observed when a blind transfer placed from the callee
channel of a test call failed. While the problem I am solving
here is exactly the same as what was reported in issue #13584,
the difference is that this fix I am applying is trunk-only.
Issue #13584 was reported against the 1.4 branch, and my tests of
1.4's blind transfers appear to work fine.
2008-10-01 17:26 +0000 [r145487] Leif Madsen <lmadsen@digium.com>
* contrib/scripts/realtime_pgsql.sql, /: Merged revisions 145479
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r145479 | lmadsen | 2008-10-01 13:18:30 -0400 (Wed, 01 Oct 2008)
| 6 lines Update the realtime_pgsql.sql script to create the
setinterfacevar column. (closes issue #13549) Reported by: fiddur
........
2008-10-01 15:44 +0000 [r145428] Tilghman Lesher <tlesher@digium.com>
* apps/app_sms.c: Initializing buffer prevents a segfault when
arguments are incomplete. (closes issue #13471) Reported by:
alecdavis Patches: 20080916__bug13471.diff.txt uploaded by
Corydon76 (license 14) Tested by: alecdavis
2008-10-01 14:44 +0000 [r145381] Mark Michelson <mmichelson@digium.com>
* Makefile: Too many times have I mistyped the word 'install' as
'isntall' Now this typo shall no longer be a problem since 'make
isntall' just builds the 'install' target.
2008-10-01 12:29 +0000 [r145329] Russell Bryant <russell@digium.com>
* CHANGES: tabs to spaces
2008-09-30 22:21 +0000 [r145249] Jeff Peeler <jpeeler@digium.com>
* channels/chan_sip.c: (closes issue #13337) Reported by: pj Tested
by: pj Set transport to SIP_TRANSPORT_UDP mode if not specified
which fixes calls to get_transport returning UNKNOWN.
2008-09-30 21:32 +0000 [r145226] Russell Bryant <russell@digium.com>
* channels/chan_sip.c, CHANGES: Add support for call pickup on Snom
phones. Asterisk now includes a magic call-id in the dialog-info
event package used with extension state subscriptions on Snom
phones. Then, when the phone sends an INVITE with Replaces for
the special callid, Asterisk will perform a pickup on the
extension that was subscribed to. The original code on this issue
was submitted by xylome. However, contributions have been made by
(at least) mgernoth and pkempgen. The final patch was written by
seanbright, and includes the necessary logic to allow this work
in a technology independent way. (closes issue #5014) Reported
by: xylome Patches: issue5014-trunk.diff uploaded by seanbright
(license 71)
2008-09-30 21:00 +0000 [r145200] Richard Mudgett <rmudgett@digium.com>
* channels/misdn/isdn_lib.h, doc/tex/misdn.tex,
channels/chan_misdn.c, channels/misdn/isdn_lib.c: * Miscellaneous
formatting changes to make v1.4 and trunk more merge compatible
in the mISDN area. channels/chan_misdn.c * Eliminated redundant
code in cb_events() EVENT_SETUP
2008-09-28 23:32 +0000 [r145121] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_unistim.c, res/res_config_pgsql.c,
apps/app_meetme.c, res/ais/clm.c, res/res_limit.c,
main/taskprocessor.c, channels/chan_console.c, apps/app_queue.c,
channels/chan_oss.c, main/astobj2.c, main/cli.c,
channels/chan_dahdi.c, main/manager.c, channels/chan_misdn.c,
channels/chan_features.c, res/res_agi.c, channels/chan_h323.c,
res/ais/evt.c, res/res_config_ldap.c, apps/app_mixmonitor.c,
res/res_clioriginate.c: Merge the cli_cleanup branch. This work
is done by lmadsen, junky and mvanbaak during AstriDevCon. This
is the second audit the CLI got, and this time lmadsen made sure
he had _ALL_ modules loaded that have CLI commands in them.
2008-09-28 21:39 +0000 [r145076] Tilghman Lesher <tlesher@digium.com>
* res/res_config_pgsql.c: Change several improper "sizeof" to
"strlen", as sizeof in that context would incorrectly use the
size of a pointer, rather than the length of a string. (Closes
issue #13574)
2008-09-28 17:08 +0000 [r145027] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_dahdi.c: rename chandup() and clarify its usage
2008-09-27 16:17 +0000 [r144949-144951] Kevin P. Fleming <kpfleming@digium.com>
* utils/Makefile: remove incorrect comment
* agi/Makefile, utils/Makefile, include/asterisk/astmm.h: fix bugs
caused by r144949 when MALLOC_DEBUG is defined
* include/asterisk.h, /, main/Makefile, main/ast_expr2.y,
Makefile.moddir_rules, utils/astman.c, main/ast_expr2.c,
Makefile, utils/Makefile, main/ast_expr2f.c, pbx/pbx_ael.c,
main/astmm.c, utils/ael_main.c, main/stdtime/localtime.c,
utils/extconf.c, main/ast_expr2.fl: Merged revisions
144924-144925 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r144924 | kpfleming | 2008-09-27 10:00:48 -0500 (Sat, 27 Sep
2008) | 6 lines improve header inclusion process in a few small
ways: - it is no longer necessary to forcibly include
asterisk/autoconfig.h; every module already includes asterisk.h
as its first header (even before system headers), which serves
the same purpose - astmm.h is now included by asterisk.h when
needed, instead of being forced by the Makefile; this means
external modules will build properly against installed headers
with MALLOC_DEBUG enabled - simplify the usage of some of these
headers in the AEL-related stuff in the utils directory ........
r144925 | kpfleming | 2008-09-27 10:13:30 -0500 (Sat, 27 Sep
2008) | 2 lines fix some minor issues with rev 144924 ........
2008-09-27 00:49 +0000 [r144879] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_dahdi.c, apps/app_queue.c: fix a couple of CLI
commands that did not have a help description.
2008-09-26 23:12 +0000 [r144829] Joshua Colp <jcolp@digium.com>
* configs/rtp.conf.sample: Update documentation to include default
setting. This is for you jtodd!
2008-09-26 18:02 +0000 [r144482-144681] Steve Murphy <murf@digium.com>
* pbx/pbx_lua.c: (closes issue #13564) Reported by: mnicholson
Patches: pbx_lua9.diff uploaded by mnicholson (license 96) Many
thanks to Matt for his upgrade to the lua dialplan option! the
Description from the bug: This patch adds a stack trace to errors
encountered while executing lua extensions. The patch also
handles out of memory errors reported by lua.
* main/pbx.c, /: Merged revisions 144677 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r144677 | murf | 2008-09-26 11:47:13 -0600 (Fri, 26 Sep 2008) |
12 lines (closes issue #13563) Reported by: mnicholson Patches:
found1.diff uploaded by mnicholson (license 96) This patch was
mainly meant to apply to trunk and 1.6.x, but I'm applying it to
1.4 also, which should be a perfectly harmless fix to the vast
majority of users who are not using external switches, but the
few who might be affected will not have to go to the pain of
filing a bug report. ........
* utils/build-extensions-conf.lua (removed): Matt suggests we
remove utils/build-extensions-conf.lua, as per bug 12961, it is
no longer necessary.
* main/pbx.c, funcs/func_cut.c, channels/chan_oss.c,
apps/app_playback.c: (closes issue #13557) Reported by:
nickpeirson The user attached a patch, but the license is not yet
recorded. I took the liberty of finding and replacing ALL index()
calls with strchr() calls, and that involves more than just
main/pbx.c; chan_oss, app_playback, func_cut also had calls to
index(), and I changed them out. 1.4 had no references to index()
at all.
* pbx/pbx_lua.c: (closes issue #13559) Reported by: mnicholson
Patches: pbx_lua8.diff uploaded by mnicholson (license 96)
* pbx/pbx_lua.c, configs/extensions.lua.sample,
include/asterisk/hashtab.h: I added a little verbage to hashtab
for the hashtab_destroy func. It was pretty sparsely documented.
This update fleshes out the pbx_lua module, to add the switch
statements to the extensions in the extensions.lua file, as well
as removing them when the module is unloaded. Many thanks to Matt
Nicholson for his fine contribution!
* pbx/pbx_lua.c: (closes issue #13558) Reported by: mnicholson
Considering that the example extensions.lua used nothing but
["12345"] notation, and that the resulting error message: [Sep 24
17:01:16] ERROR[12393]: pbx_lua.c:1204 exec: Error executing lua
extension: attempt to call a nil value is not very informative as
to the nature of the problem, I think this bug fix is a big win!
2008-09-25 01:46 +0000 [r144357] Tilghman Lesher <tlesher@digium.com>
* /: Recorded merge of revisions 144356 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r144356 | tilghman | 2008-09-24 20:44:47 -0500 (Wed, 24 Sep 2008)
| 6 lines Backport Hebrew language to voicemail. (closes issue
#13155) Reported by: greenfieldtech Patches:
voicemail-hebrew-patch-1.4-SVN.c.patch uploaded by greenfieldtech
(license 369) ........
2008-09-24 22:05 +0000 [r144314] Doug Bailey <dbailey@digium.com>
* res/res_phoneprov.c: Blanch the 404 error message for those with
no sense of humor
2008-09-24 08:42 +0000 [r144257] Christian Richter <christian.richter@beronet.com>
* channels/chan_misdn.c, /: Merged revisions 144238 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r144238 | crichter | 2008-09-24 10:20:52 +0200 (Mi, 24
Sep 2008) | 1 line improved helptext of misdn_set_opt. ........
2008-09-24 06:43 +0000 [r144199] Tilghman Lesher <tlesher@digium.com>
* funcs/func_curl.c: Create a 'hashcompat' option that permits the
results of a CURL() able to be passed directly into the HASH()
function. Requested via the -users list, and committed at
Astricon in the Code Zone.
2008-09-23 23:33 +0000 [r144149] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Fix a conflict in flag values
2008-09-23 16:52 +0000 [r144067] Steve Murphy <murf@digium.com>
* /, main/features.c: Merged revisions 144066 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r144066 | murf | 2008-09-23 10:41:49 -0600 (Tue, 23 Sep 2008) |
29 lines (closes issue #13489) Reported by: DougUDI Tested by:
murf (closes issue #13490) Reported by: seanbright Tested by:
murf (closes issue #13467) Reported by: edantie Tested by: murf,
edantie, DougUDI This crash happens because we are unsafely
handling old pointers. The channel whose cdr is being handled,
has been hung up and destroyed already. I reorganized the code a
bit, and tried not to lose the fork-cdr-chain concepts of the
previous code. I now verify that the 'previous' channel (the
channel we had when the bridge was started), still exists, by
looking it up by name in the channel list. I also do not try to
reset the CDR's of channels involved in bridges. Testing shows it
solves the crash problem, and should not negatively impact
previous fixes involving CDR's generated during/after blind
transfers. (The reason we need to reset the CDR's on the
"beginning" channels in the first place). ........
2008-09-23 15:37 +0000 [r144025] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: When a promiscuous redirect contained both a
user and host portion in the Contact URI and specifies a
transport, the parsing done in parse_moved_contact resulted in a
malformed URI. This commit fixes the parsing so that a proper
Dial string may be formed when the forwarded call is placed.
(closes issue #13523) Reported by: mattdarnell Patches:
13523v2.patch uploaded by putnopvut (license 60) Tested by:
mattdarnell
2008-09-22 22:50 +0000 [r143904] Sean Bright <sean.bright@gmail.com>
* /, formats/format_pcm.c: Merged revisions 143903 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r143903 | seanbright | 2008-09-22 18:49:00 -0400 (Mon,
22 Sep 2008) | 8 lines Use the advertised header size in .au
files instead of just assuming they are 24 bytes (the minimum).
(closes issue #13450) Reported by: jamessan Patches:
pcm-header.diff uploaded by jamessan (license 246) ........
2008-09-21 09:53 +0000 [r143799-143843] Michiel van Baak <michiel@vanbaak.info>
* doc/tex/privacy.tex: fix privacymanager example so it shows how
to use the PRIVACYMRGSTATUS variable
* doc/tex/privacy.tex: document the new context argument for
privacymanager so people can do pattern matching on the input
* doc/tex/privacy.tex: fix privacy documentation. We no longer do
priority jumping +101
* channels/chan_skinny.c: make 'module unload chan_skinny.so'
actually work. (closes issue #13524) Reported by: wedhorn
Patches: unload.diff uploaded by wedhorn (license 30) With small
tweak by me to prevent a crash
2008-09-20 00:52 +0000 [r143737] Sean Bright <sean.bright@gmail.com>
* /, contrib/scripts/vmail.cgi: Merged revisions 143736 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r143736 | seanbright | 2008-09-19 20:50:10 -0400 (Fri, 19 Sep
2008) | 9 lines Make vmail.cgi work with mailboxes defined in
users.conf, too. (closes issue #13187) Reported by: netvoice
Patches: 20080911__bug13187.diff.txt uploaded by Corydon76
(license 14) (Slightly modified to take alchamist's comments on
mantis into account) Tested by: msales, alchamist, seanbright
........
2008-09-19 21:41 +0000 [r143697] Steve Murphy <murf@digium.com>
* /: This blocks 143674 from trunk; it appears to already done in
trunk, since ast_odbc_direct_execute creates a new stmt for each
attempt.
2008-09-19 15:43 +0000 [r143609] Mark Michelson <mmichelson@digium.com>
* channels/chan_agent.c: We should only unsubscribe to the device
state event subscription if we have previously subscribed.
Otherwise a segfault will occur. (closes issue #13476) Reported
by: jonnt Patches: 13476.patch uploaded by putnopvut (license 60)
Tested by: jonnt
2008-09-18 23:41 +0000 [r143559] Steve Murphy <murf@digium.com>
* /, channels/chan_sip.c: Merged revisions 143534 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r143534 | murf | 2008-09-18 16:11:51 -0600 (Thu, 18 Sep 2008) | 1
line A micro-fix, in sip_park_thread, where d is freed before the
func is done using it. ........
2008-09-17 20:57 +0000 [r143405] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_voicemail.c: Merged revisions 143404 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r143404 | tilghman | 2008-09-17 15:55:47 -0500 (Wed, 17
Sep 2008) | 6 lines When callerid is blank, we want to use
"unknown caller" in those cases, too. (closes issue #13486)
Reported by: tomo1657 Patches: 20080917__bug13486.diff.txt
uploaded by Corydon76 (license 14) ........
2008-09-17 20:25 +0000 [r143340-143400] Mark Michelson <mmichelson@digium.com>
* main/astmm.c: If attempting to free a NULL pointer when
MALLOC_DEBUG is set, don't bother searching for a region to free,
just immediately exit. This has the dual benefit of suppressing a
warning message about freeing memory at (nil) and of optimizing
the free() replacement by not having to do any futile searching
for the proper region to free. (closes issue #13498) Reported by:
pj Patches: 13498.patch uploaded by putnopvut (license 60) Tested
by: pj
* /, main/rtp.c: Merged revisions 143337 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r143337 | mmichelson | 2008-09-17 13:24:15 -0500 (Wed, 17 Sep
2008) | 6 lines Allow for "G.729" if offered in an SDP even
though it is not RFC 3551 compliant. Some Cisco switches will
send this in an SDP, and it doesn't hurt to be able to accept
this. ........
2008-09-15 21:31 +0000 [r143141] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_iax2.c: Merged revisions 143140 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r143140 | tilghman | 2008-09-15 16:29:32 -0500 (Mon, 15
Sep 2008) | 6 lines Set the raw formats at the same time as the
other formats. (closes issue #13240) Reported by: jvandal
Patches: 20080813__bug13240.diff.txt uploaded by Corydon76
(license 14) ........
2008-09-14 22:16 +0000 [r143082] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_skinny.c: plug a couple of memleaks in chan_skinny.
(closes issue #13452) Reported by: pj Patches: memleak5.diff
uploaded by wedhorn (license 30) Tested by: wedhorn, pj, mvanbaak
(closes issue #13294) Reported by: pj
2008-09-13 14:15 +0000 [r143034] Sean Bright <sean.bright@gmail.com>
* apps/app_osplookup.c: Everytime a compile fails, a puppy dies.
2008-09-13 13:54 +0000 [r142992-143031] Tilghman Lesher <tlesher@digium.com>
* apps/app_dial.c, channels/chan_iax2.c, channels/iax2-parser.c:
Repair IAXVAR implementation so that it works again (regression?)
(closes issue #13354) Reported by: adomjan Patches:
20080828__bug13354.diff.txt uploaded by Corydon76 (license 14)
20080829__bug13354__1.6.0.diff.txt uploaded by Corydon76 (license
14) Tested by: Corydon76, adomjan
* channels/chan_unistim.c, main/udptl.c, apps/app_meetme.c,
res/res_snmp.c, codecs/codec_adpcm.c, res/res_phoneprov.c,
codecs/codec_gsm.c, apps/app_alarmreceiver.c,
channels/chan_gtalk.c, res/res_http_post.c,
res/res_musiconhold.c, channels/chan_iax2.c, apps/app_followme.c,
res/res_jabber.c, main/enum.c, res/res_config_sqlite.c,
main/config.c, main/loader.c, main/cdr.c, channels/chan_dahdi.c,
channels/chan_phone.c, res/res_smdi.c, main/manager.c,
funcs/func_config.c, apps/app_osplookup.c,
channels/chan_skinny.c, funcs/func_odbc.c, main/features.c,
apps/app_minivm.c, main/http.c, channels/chan_alsa.c,
apps/app_amd.c, apps/app_directory.c, res/res_config_ldap.c,
apps/app_rpt.c, channels/chan_mgcp.c, codecs/codec_lpc10.c,
res/res_config_pgsql.c, main/dnsmgr.c, codecs/codec_g722.c,
channels/chan_sip.c, apps/app_festival.c, codecs/codec_speex.c,
codecs/codec_alaw.c, res/res_adsi.c, include/asterisk/config.h,
channels/chan_agent.c, codecs/codec_g726.c,
channels/chan_console.c, apps/app_queue.c, channels/chan_oss.c,
main/rtp.c, apps/app_playback.c, channels/chan_jingle.c,
channels/chan_h323.c, codecs/codec_ulaw.c, codecs/codec_dahdi.c,
res/res_indications.c, main/asterisk.c, res/res_odbc.c,
main/dsp.c, apps/app_voicemail.c: Create a new config file
status, CONFIG_STATUS_FILEINVALID for differentiating when a file
is invalid from when a file is missing. This is most important
when we have two configuration files. Consider the following
example: Old system: sip.conf users.conf Old result New result
======== ========== ========== ========== Missing Missing SIP
doesn't load SIP doesn't load Missing OK SIP doesn't load SIP
doesn't load Missing Invalid SIP doesn't load SIP doesn't load OK
Missing SIP loads SIP loads OK OK SIP loads SIP loads OK Invalid
SIP loads incompletely SIP doesn't load Invalid Missing SIP
doesn't load SIP doesn't load Invalid OK SIP doesn't load SIP
doesn't load Invalid Invalid SIP doesn't load SIP doesn't load So
in the case when users.conf doesn't load because there's a typo
that disrupts the syntax, we may only partially load users,
instead of failing with an error, which may cause some calls not
to get processed. Worse yet, the old system would do this with no
indication that anything was even wrong. (closes issue #10690)
Reported by: dtyoo Patches: 20080716__bug10690.diff.txt uploaded
by Corydon76 (license 14)
2008-09-12 22:24 +0000 [r142929] Jeff Peeler <jpeeler@digium.com>
* channels/chan_local.c, /: Merged revisions 142927 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r142927 | jpeeler | 2008-09-12 17:22:28 -0500 (Fri, 12
Sep 2008) | 6 lines (closes issue #12965) Reported by: rlsutton2
Prevents local channels from playing MOH at each other which was
causing ast_generic_bridge to loop much faster. ........
2008-09-12 20:49 +0000 [r142866] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
142865 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r142865 | tilghman | 2008-09-12 15:37:18 -0500 (Fri, 12 Sep 2008)
| 11 lines Create rules for disallowing contacts at certain
addresses, which may improve the security of various
installations. As this does not change any default behavior, it
is not classified as a direct security fix for anything within
Asterisk, but may help PBX admins better secure their SIP
servers. (closes issue #11776) Reported by: ibc Patches:
20080829__bug11776.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76, blitzrage ........
2008-09-12 18:22 +0000 [r142808] Michiel van Baak <michiel@vanbaak.info>
* /: Recorded merge of revisions 142807 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r142807 | mvanbaak | 2008-09-12 19:59:25 +0200 (Fri, 12 Sep 2008)
| 2 lines fix copyright year range ........
2008-09-12 16:54 +0000 [r142741-142748] Tilghman Lesher <tlesher@digium.com>
* main/app.c: When checking for an encoded character, make sure the
string isn't blank, first. (Closes issue #13470)
* /, apps/app_voicemail.c: Merged revisions 142744 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r142744 | tilghman | 2008-09-12 11:38:02 -0500 (Fri, 12
Sep 2008) | 4 lines Missing merge from 1.2 fixes errant exit on
DTMF, only when language is Italian (cf commit 34242) (Closes
issue #7353) ........
* /, main/file.c: Merged revisions 142740 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r142740 | tilghman | 2008-09-12 11:27:32 -0500 (Fri, 12 Sep 2008)
| 4 lines Don't return a free'd pointer, when a file cannot be
opened. (closes issue #13462) Reported by: wackysalut ........
2008-09-12 04:50 +0000 [r142676] Steve Murphy <murf@digium.com>
* apps/app_dial.c, main/pbx.c, /, main/features.c,
include/asterisk/channel.h, apps/app_queue.c: Merged revisions
142675 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r142675 | murf | 2008-09-11 22:29:34 -0600 (Thu, 11 Sep 2008) |
29 lines Tested by: sergee, murf, chris-mac, andrew, KNK This is
a "second attempt" to restore the previous "endbeforeh" behavior
in 1.4 and up. In order to capture information concerning all the
legs of transfers in all their infinite combinations, I was
forced to this particular solution by a chain of logical
necessities, the first being that I was not allowed to rewrite
the CDR mechanism from the ground up! This change basically
leaves the original machinery alone, which allows IVR and local
channel type situations to generate CDR's as normal, but a
channel flag can be set to suppress the normal running of the h
exten. That flag would be set by the code that runs the h exten
from the ast_bridge_call routine, to prevent the h exten from
being run twice. Also, a flag in the ast_bridge_config struct
passed into ast_bridge_call can be used to suppress the running
of the h exten in that routine. This would happen, for instance,
if you use the 'g' option in the Dial app. Running this routine
'early' allows not only the CDR() func to be used in the h
extension for reading CDR variables, but also allows them to be
modified before the CDR is posted to the backends. While I dearly
hope that this patch overcomes all problems, and introduces no
new problems, reality suggests that surely someone will have
problems. In this case, please re-open 13251 (or 13289), and
we'll see if we can't fix any remaining issues. ** trunk note:
some code to suppress the h exten being run from app_queue was
added; for the 'continue' option available only in trunk/1.6.x.
........
2008-09-12 00:49 +0000 [r142635] Sean Bright <sean.bright@gmail.com>
* cdr/cdr_adaptive_odbc.c: Build under dev-mode
2008-09-11 23:12 +0000 [r142576] Steve Murphy <murf@digium.com>
* /, main/features.c: Merged revisions 142575 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r142575 | murf | 2008-09-11 16:55:49 -0600 (Thu, 11 Sep 2008) |
20 lines (closes issue #13364) Reported by: mdu113 Well,
fundamentally, the problems revealed in 13364 are because of the
ForkCDR call that is done before the dial. When the bridge is in
place, it's dealing with the first (and wrong) cdr in the list.
So, I wrote a little func to zip down to the first non-locked cdr
in the chain, and thru-out the ast_bridge_call, these results are
used instead of raw chan->cdr and peer->cdr pointers. This
shouldn't affect anyone who isn't forking cdrs before a dial, and
should correct the cdr's of those that do. So, this change ends
up correcting the dstchannel and userfield; the disposition was
fixed by a previous patch, it was OK coming into this problem.
........
2008-09-11 21:45 +0000 [r142536] Tilghman Lesher <tlesher@digium.com>
* cdr/cdr_adaptive_odbc.c, configs/cdr_adaptive_odbc.conf.sample:
Add usegmtime, as per the recent -users list discussion, and also
add my explanation to the file, since that additional text helps
people understand the concept.
2008-09-10 22:11 +0000 [r142475] Steve Murphy <murf@digium.com>
* /, main/features.c: Merged revisions 142474 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r142474 | murf | 2008-09-10 15:58:17 -0600 (Wed, 10 Sep 2008) |
30 lines (closes issue #12318) Reported by: krtorio I made a
small change to the code that handles local channel situations.
In that code, I copy the answer time from the peer cdr, to the
bridge_cdr, but I wasn't also copying the disposition from the
peer cdr. So, Now I copy the disposition, and I've tested against
these cases: 1. phone 1 never answers the phone; no cdr is
generated at all. this should show up as a manager command
failure or something. 2. phone 2 never answers. CDR is generated,
says NO ANSWER 3. phone 2 is busy. CDR is generated, says BUSY 4.
phone 2 answers: CDR is generated, times are correct; disposition
is ANSWERED, which is correct. The start time is the time that
the manager dialed the first phone. The answer time is the time
the second phone picks up. I purposely left the cid and src
fields blank; since this call really originates from the manager,
there is no 'easy' data to put in these fields. If you feel
strongly that these fields should be filled in, re-open this bug
and I'll dig further. ........
2008-09-10 19:09 +0000 [r142417] Sean Bright <sean.bright@gmail.com>
* /, configure, acinclude.m4: Merged revisions 142416 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r142416 | seanbright | 2008-09-10 15:05:46 -0400 (Wed,
10 Sep 2008) | 9 lines Fix detection of PWLIB and OpenH323
version when spacing in the headers isn't consistent. (closes
issue #13426) Reported by: bamby Patches: detect_openh323.diff
uploaded by bamby (license 430) (Modified by me to use sed
instead of tr) ........
2008-09-10 16:55 +0000 [r142359] Tilghman Lesher <tlesher@digium.com>
* /, sounds/Makefile: Merged revisions 142358 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r142358 | tilghman | 2008-09-10 11:54:29 -0500 (Wed, 10 Sep 2008)
| 2 lines Publish new extra sounds version. ........
2008-09-10 16:41 +0000 [r142318-142355] Russell Bryant <russell@digium.com>
* /, main/sched.c: Merged revisions 142354 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r142354 | russell | 2008-09-10 11:39:53 -0500 (Wed, 10 Sep 2008)
| 7 lines It is a normal situation that a task gets put in the
scheduler that should run as soon as possible. Accept "0" as an
acceptable time to run, and also treat negative as "run now", and
don't print a debug message about it. (inspired by a message
asking about the "request to schedule in the past" debug message
on the -dev list) ........
* CHANGES: Move last change to CHANGES up to the 1.6.2 section
2008-09-09 22:08 +0000 [r142280] Philippe Sultan <philippe.sultan@gmail.com>
* configs/jabber.conf.sample, CHANGES, res/res_jabber.c: Disable
autoprune by default. (closes issue #13411) Reported by: caio1982
Patches: res_jabber_autoprune1.diff uploaded by caio1982 (license
22) Tested by: caio1982
2008-09-09 19:16 +0000 [r142219] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 142218 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r142218 | mmichelson | 2008-09-09 14:15:28 -0500 (Tue, 09 Sep
2008) | 14 lines Make sure that the branch sent in CANCEL
requests matches the branch of the INVITE it is cancelling.
(closes issue #13381) Reported by: atca_pres Patches:
13381v2.patch uploaded by putnopvut (license 60) Tested by:
atca_pres (closes issue #13198) Reported by: rickead2000 Tested
by: rickead2000 ........
2008-09-09 17:30 +0000 [r142181] Richard Mudgett <rmudgett@digium.com>
* main/callerid.c: Cleaned up comment
2008-09-09 17:15 +0000 [r142080-142146] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: This is the trunk version of the patch to close
issue 12979. The difference between this and the 1.6.0 and 1.6.1
versions is that this is a much more invasive change. With this,
we completely get rid of the interfaces list, along with all its
helper functions. Let me take a moment to say that this change
personally excites me since it may mean huge steps forward
regarding proper lock order in app_queue without having to strew
seemingly unnecessary locks all over the place. It also results
in a huge reduction in lines of code and complexity. Way to go
Brett! (closes issue #12979) Reported by: sigxcpu Patches:
20080710_issue12979_queue_custom_state_interface_trunk_2.diff
uploaded by bbryant (license 36) Tested by: sigxcpu, putnopvut
* /, channels/chan_sip.c: Merged revisions 142079 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r142079 | mmichelson | 2008-09-09 11:19:17 -0500 (Tue, 09 Sep
2008) | 21 lines When determining if codecs used by SIP peers
allow the media to be natively bridged, use the jointcapability
instead of the peercapability. It seems that the intent of using
the peercapability was to expand the choice of codecs for the
call to increase the chances of being able to native bridge the
channels. The problem is that if a codec were settled on for the
native bridge and that wasn't a codec that was configured to be
used by Asterisk for that peer, then Asterisk would send a
REINVITE with no codecs in the SDP which is a bug no matter how
you slice it. (closes issue #13076) Reported by: ramonpeek
Patches: 13076.patch uploaded by putnopvut (license 60) Tested
by: tbelder ........
2008-09-09 15:44 +0000 [r142064] Russell Bryant <russell@digium.com>
* /, main/features.c: Merged revisions 142063 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r142063 | russell | 2008-09-09 10:40:24 -0500 (Tue, 09 Sep 2008)
| 5 lines Ensure that the stored CDR reference is still valid
after the bridge before poking at it. Also, keep the channel
locked while messing with this CDR. (fixes crashes reported in
issue #13409) ........
2008-09-09 12:34 +0000 [r142000] Bradley Latus <brad.latus@gmail.com>
* include/asterisk/astobj2.h: Minor fix to doco
2008-09-09 12:32 +0000 [r141995-141998] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Use ast_debug for debug messages. I was
wondering why debug messages weren't showing up when I had set
the debug level high for just app_queue.c. It's because we were
only checking the global option_debug variable instead of using
the awesome macro which checks both the global and file-specific
value
* channels/chan_oss.c: Fix a memory leak in chan_oss (closes issue
#13311) Reported by: eliel Patches: chan_oss.c.patch uploaded by
eliel (license 64)
2008-09-09 01:47 +0000 [r141949] Russell Bryant <russell@digium.com>
* main/channel.c: Modify ast_answer() to not hold the channel lock
while calling ast_safe_sleep() or when calling ast_waitfor().
These are inappropriate times to hold the channel lock. This is
what has caused "could not get the channel lock" messages from
chan_sip and has likely caused a negative impact on performance
results of SIP in Asterisk 1.6. Thanks to file for pointing out
this section of code. (closes issue #13287) (closes issue #13115)
2008-09-08 23:00 +0000 [r141810-141906] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Optimization: The only reason we should check
member status is if the queue has a joinempty or a leavewhenempty
setting which could cause the caller to not join the queue or
exit the queue. Prior to this patch, we could potentially
traverse the entire queue's member list for no reason since even
if the members are currently not available in some way we're
going to let the caller join the queue anyway.
* channels/chan_sip.c: Um, apparently I didn't actually finish
merging before committing. Bad bad bad
* /, channels/chan_sip.c: Merged revisions 141809 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r141809 | mmichelson | 2008-09-08 16:10:10 -0500 (Mon, 08 Sep
2008) | 14 lines Fix pedantic mode of chan_sip to only check the
remote tag of an endpoint once a dialog has been confirmed. Up
until that point, it is possible and legal for the far-end to
send provisional responses with a different To: tag each time.
With this patch applied, these provisional messages will not
cause a matching problem. (closes issue #11536) Reported by: ibc
Patches: 11536v2.patch uploaded by putnopvut (license 60)
........
2008-09-08 21:05 +0000 [r141807] Russell Bryant <russell@digium.com>
* main/pbx.c, /: Merged revisions 141806 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r141806 | russell | 2008-09-08 16:02:36 -0500 (Mon, 08 Sep 2008)
| 7 lines When doing an async goto, detect if the channel is
already in the middle of a masquerade. This can happen when
chan_local is trying to optimize itself out. If this happens,
fail the async goto instead of bursting into flames. (closes
issue #13435) Reported by: geoff2010 ........
2008-09-08 20:18 +0000 [r141745] Jason Parker <jparker@digium.com>
* Makefile, /, redhat (removed): Merged revisions 141741 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r141741 | qwell | 2008-09-08 15:15:42 -0500 (Mon, 08 Sep 2008) |
8 lines Remove RPM package targets from Makefile (and all
associated parts). This has never worked in 1.4, and we decided
that it makes no sense to be done here. There are many distros
out there that already have "proper" spec files that can be
(re)used. Closes issue #13113 Closes issue #10950 Closes issue
#10952 ........
2008-09-08 17:13 +0000 [r141682] Sean Bright <sean.bright@gmail.com>
* build_tools/make_buildopts_h: Quote the arguments to grep so that
sh on various platforms doesn't choke on the special characters
(like ^). (closes issue #13417) Reported by: dougm Patches:
13417.make_buildopts_h.patch uploaded by seanbright (license 71)
Tested by: dougm
2008-09-07 00:04 +0000 [r141626] Michiel van Baak <michiel@vanbaak.info>
* funcs/func_curl.c: make func_curl.c compile under devmode.
2008-09-06 20:19 +0000 [r141566] Steve Murphy <murf@digium.com>
* /, channels/chan_sip.c: Merged revisions 141565 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r141565 | murf | 2008-09-06 14:13:16 -0600 (Sat, 06 Sep 2008) | 1
line This fix comes from Joshua Colp The Brilliant, who, given
the trace, came up with a solution. This will most likely will
close 13235 and 13409. I'll wait till Monday to verify, and then
close these bugs. ........
2008-09-06 15:40 +0000 [r141504-141507] Tilghman Lesher <tlesher@digium.com>
* funcs/func_curl.c: Get rid of the casts that cause warnings on
OpenBSD. The compiler is errantly detecting warnings when we
redefine a structure each time it is used, even though the
structure is identical. Reported by: mvanbaak, via #asterisk-dev
* /, res/res_agi.c: Merged revisions 141503 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r141503 | tilghman | 2008-09-06 10:23:42 -0500 (Sat, 06 Sep 2008)
| 4 lines Reverting behavior change (AGI should not exit non-zero
on SUCCESS) (closes issue #13434) Reported by: francesco_r
........
2008-09-06 12:03 +0000 [r141464] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_sip.c, channels/chan_iax2.c, main/cli.c: Some fixes
to autocompletion in some commands. Changes applied by this
patch: - Fix autocompletion in 'sip prune realtime', sip peers
where never auto completed. Now we complete this command with:
'sip prune realtime peer' -> all | like | sip peers Also I have
modified the syntax in the usage, was wrong... - Pass
ast_cli_args->argv and ast_cli_args->argc while running
autocompletion on CLI commands (CLI_GENERATE). With this we avoid
comparisons on ast_cli_args->line like this: strcasestr(a->line,
" description") strcasestr(a->line, "descriptions ")
strcasestr(a->line, "realtime peer"), and so on.. Making the code
more confusing (check the spaces in description!). The only thing
we must be sure is to first check a->pos or a->argc. - Fix 'iax2
prune realtime' autocompletion, now we autocomplete this command
with 'all' & 'iax2 peers', check a look that iax2 peers where all
the peers, now only the ones in the cache.. (closes issue #13133)
Reported by: eliel Patches: clichanges.patch uploaded by eliel
(license 64)
2008-09-05 22:03 +0000 [r141367-141425] Mark Michelson <mmichelson@digium.com>
* funcs/func_curl.c: Fix func_curl compilation
* /, channels/chan_agent.c: Merged revisions 141366 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r141366 | mmichelson | 2008-09-05 16:10:32 -0500 (Fri,
05 Sep 2008) | 7 lines Agent's should not try to call a channel's
indicate callback if the channel has been hung up. It will likely
crash otherwise ABE-1159 ........
2008-09-05 19:12 +0000 [r141328] Tilghman Lesher <tlesher@digium.com>
* funcs/func_curl.c, CHANGES: Add the CURLOPT dialplan function,
which permits setting various options for use with the CURL
dialplan function. (closes issue #12920) Reported by: davevg
Patches: 20080904__bug12920.diff.txt uploaded by Corydon76
(license 14) Tested by: Corydon76, davevg
2008-09-05 14:18 +0000 [r141115-141157] Steve Murphy <murf@digium.com>
* main/channel.c, /: Merged revisions 141156 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r141156 | murf | 2008-09-05 08:15:43 -0600 (Fri, 05 Sep 2008) | 1
line A small change to prevent double-posting of CDR's; thanks to
Daniel Ferrer for bringing it to our attention ........
* pbx/ael/ael-test/ref.ael-vtest25 (added), /,
pbx/ael/ael-test/ael-vtest25/extensions.ael,
pbx/ael/ael-test/ael-vtest25 (added), res/ael/ael_lex.c,
pbx/ael/ael-test/ref.ael-test6, res/ael/ael.flex: Merged
revisions 141094 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r141094 | murf | 2008-09-04 17:15:07 -0600 (Thu, 04 Sep 2008) |
70 lines (closes issue #13357) Reported by: pj Tested by: murf
(closes issue #13416) Reported by: yarns Tested by: murf If you
find this message overly verbose, relax, it's probably not meant
for you. This message is meant for probably only two people in
the whole world: me, or the poor schnook that has to maintain
this code because I'm either dead or unavailable at the moment.
This fix solves two reports, both having to do with embedding a
function call in a ${} construct. It was tricky because the
funccall syntax has parenthesis () in it. And up till now, the
'word' token in the flex stuff didn't allow that, because it
would tend to steal the LP and RP tokens. To be truthful, the
"word" token was the trickiest, most unstable thing in the whole
lexer. I was lucky it made this long without complaints. I had to
choose every character in the pattern with extreme care, and I
knew that someday I'd have to revisit it. Well, the day has come.
So, my brilliant idea (and I'm being modest), was to use the
surrounding ${} construct to make a state machine and capture
everything in it, no matter what it contains. But, I have to now
treat the word token like I did with comments, in that I turn the
whole thing into a state-machine sort of spec, with new contexts
"curlystate", "wordstate", and "brackstate". Wait a minute,
"brackstate"? Yes, well, it didn't take very many regression
tests to point out if I do this for ${} constructs, I also have
to do it with the $[] constructs, too. I had to create a separate
pcbstack2 and pcbstack3 because these constructs can occur inside
macro argument lists, and when we have two state machines
operating on the same structures we'd get problems otherwise. I
guess I could have stopped at pcbstack2 and had the brackstate
stuff share it, but it doesn't hurt to be safe. So, the pcbpush
and pcbpop routines also now have versions for "2" and "3". I had
to add the {KEYWORD} construct to the initial pattern for "word",
because previously word would match stuff like "default7",
because it was a longer match than the keyword "default". But,
not any more, because the word pattern only matches only one or
two characters now, and it will always lose. So, I made it the
winner again by making an optional match on any of the keywords
before it's normal pattern. I added another regression test to
make sure we don't lose this in future edits, and had to fix just
one regression, where it no longer reports a 'cascaded' error,
which I guess is a plus. I've given some thought as to whether to
apply these fixes to 1.4 and the 1.6.x releases, vs trunk; I
decided to put it in 1.4 because one of the bug reports was
against 1.4; and it is unexpected that AEL cannot handle this
situation. It actually reduced the amount of useless "cascade"
error messages that appeared in the regressions (by one line,
ehhem). There is a possible side-effect in that it does now do
more careful checking of what's in those ${} constructs, as far
as matching parens, and brackets are concerned. Some users may
find a an insidious problem and correct it this way. This should
be exceedingly rare, I hope. ........
2008-09-04 17:27 +0000 [r141039] Jeff Peeler <jpeeler@digium.com>
* /, main/features.c, res/res_agi.c: Merged revisions 141028 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r141028 | jpeeler | 2008-09-04 12:00:29 -0500 (Thu, 04 Sep 2008)
| 7 lines (closes issue #11979) Fixes multiple parking problems:
Crash when executing a park on an extension dialed by AGI due to
not returning the proper return code. Crash when using a builtin
feature that was a subset of a enabled dynamic feature. Crash due
to always hanging up the peer despite the fact that the peer was
supposed to be parked. ........
2008-09-03 20:16 +0000 [r140975] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Fix some locking order issues in app_queue.
This was brought up by atis on IRC a while ago.
2008-09-03 18:06 +0000 [r140938] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_skinny.c, CHANGES: Added 'skinny show lines
verbose' This will print the subs and their status for every line
(if any). wedhorn did most of the work with his patch which
introduced 'skinny show debug' but a discussion on IRC stated
that it should be added to 'skinny show lines' Input on the
output format by Qwell on IRC. (closes issue #13344) Reported by:
wedhorn
2008-09-03 14:41 +0000 [r140860-140887] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: Fix compilation
* /, apps/app_voicemail.c: Merged revisions 140850 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r140850 | mmichelson | 2008-09-03 09:29:15 -0500 (Wed,
03 Sep 2008) | 9 lines Fix voicemail forwarding when using ODBC
storage. (closes issue #13387) Reported by: moliveras Patches:
13387.patch uploaded by putnopvut (license 60) Tested by:
putnopvut, moliveras ........
2008-09-03 14:01 +0000 [r140824] Steve Murphy <murf@digium.com>
* res/ael/pval.c, main/pbx.c, res/ael/ael.tab.c, res/ael/ael.y,
res/ael/ael.tab.h: In these changes, I have added some
explanation of changes to the Set and MSet apps, so people aren't
so shocked and surprised when they upgrade from 1.4 to 1.6. Also,
for the sake of those upgrading from 1.4 to 1.6 with AEL, I
provide automatic support for the "old" way of using Set(), that
still does the exact same old thing with quotes and backslashes
and so on as 1.4 did, by having AEL compile in the use of MSet()
instead of Set(), everywhere it inserts this code. But, if the
app_set var is set to 1.6 or higher, it uses the "new",
non-evaluative Set(). This only usually happens if the user
manually inserts this into the asterisk.conf file, or runs the
"make samples" command.
2008-09-03 13:48 +0000 [r140821] Sean Bright <sean.bright@gmail.com>
* cdr/cdr_sqlite.c: Move some duplicated code into a separate
function. Also try to do some wacky stuff in the commit message,
like: a newline \n a bell \a a tab \t a format specification %p
That is all.
2008-09-03 13:41 +0000 [r140817-140820] Russell Bryant <russell@digium.com>
* main/pbx.c: Formatting change to test something on the svn server
* /, main/poll.c: Merged revisions 140816 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r140816 | russell | 2008-09-03 08:24:35 -0500 (Wed, 03 Sep 2008)
| 4 lines Don't freak out if the poll emulation receives NULL for
the pollfds array (closes issue #13307) Reported by: jcovert
........
2008-09-02 23:48 +0000 [r140752] Mark Michelson <mmichelson@digium.com>
* /, apps/app_voicemail.c: Merged revisions 140751 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r140751 | mmichelson | 2008-09-02 18:47:49 -0500 (Tue,
02 Sep 2008) | 6 lines After adding the context checking to
app_voicemail for IMAP storage, I left out a crucial place to
copy the context to the vm_state structure. This is the
correction. ........
2008-09-02 23:44 +0000 [r140691-140749] Steve Murphy <murf@digium.com>
* main/cdr.c, /: Merged revisions 140747 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r140747 | murf | 2008-09-02 17:36:56 -0600 (Tue, 02 Sep 2008) | 1
line I am turning the warnings generated in ast_cdr_free and
post_cdr into verbose level 2 messages. Really, they matter
little to end users. You either get the CDR's you wanted, or you
don't, and it is a bug. For trunk, I am going one step further.
These messages were pretty worthless even for debug, so I'm
completely removing them. ........
* main/channel.c, /: Merged revisions 140690 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r140690 | murf | 2008-09-02 16:40:13 -0600 (Tue, 02 Sep 2008) | 1
line After reconsidering, with respect to 13409, ast_cdr_detach
should be OK, better in fact, than ast_cdr_free, which generates
lots of useless warnings that will undoubtably generate
complaints. Hmmm. It doesn't hush the useless warnings, but it
does allow control of posting via the detach and post routines,
for those possible situations, where you'd want to post
single-channel cdrs. ........
* main/channel.c, main/pbx.c, /: Merged revisions 140670 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r140670 | murf | 2008-09-02 16:15:57 -0600 (Tue, 02 Sep 2008) |
14 lines (closes issue #13409) Reported by: tomaso Patches:
asterisk-1.6.0-rc2-cdrmemleak.patch uploaded by tomaso (license
564) I basically spent the day, verifying that this patch solves
the problem, and doesn't hurt in non-problem cases. Why valgrind
did not plainly reveal this leak absolutely mystifies and stuns
me. Many, many thanks to tomaso for finding and providing the
fix. ........
2008-09-02 18:15 +0000 [r140606] Sean Bright <sean.bright@gmail.com>
* /, channels/chan_iax2.c: Merged revisions 140605 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r140605 | seanbright | 2008-09-02 14:14:57 -0400 (Tue,
02 Sep 2008) | 8 lines Make sure to use the correct length of the
mohinterpret and mohsuggest buffers when copying configuration
values. (closes issue #13336) Reported by:
decryptus_proformatique Patches:
chan_iax2_mohinterpret_mohsuggest_general_settings.patch uploaded
by decryptus (license 555) ........
2008-09-02 15:11 +0000 [r140563-140566] Russell Bryant <russell@digium.com>
* codecs/codec_resample.c, apps/app_jack.c: Update instructions for
getting libresample
* res/ais/lck.c (removed), res/ais/ckpt.c (removed), res/ais/amf.c
(removed): I'm not sure how these files got to trunk (probably my
fault), but they should not be here
2008-09-02 14:41 +0000 [r140559] Sean Bright <sean.bright@gmail.com>
* channels/chan_sip.c: When a call is rejected because of
call-limit, the channel driver is behaving as expected, so we
shouldn't report it as an error. Change to LOG_NOTICE instead.
2008-08-29 17:53 +0000 [r140491] Jeff Peeler <jpeeler@digium.com>
* main/features.c, CHANGES: Added the option s to the Park
application which will silence the announcement of the parking
space number. Also, fixes the bug of just clearing the flags
instead of actually parsing the arguments to Park.
2008-08-29 17:47 +0000 [r140418-140489] Mark Michelson <mmichelson@digium.com>
* main/manager.c, res/ais/lck.c, /, channels/chan_sip.c,
funcs/func_dialgroup.c, res/res_timing_pthread.c,
main/features.c, res/res_phoneprov.c, utils/hashtest2.c,
channels/chan_console.c, main/taskprocessor.c, apps/app_queue.c,
channels/chan_iax2.c, main/config.c: Merged revisions 140488 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r140488 | mmichelson | 2008-08-29 12:34:17 -0500 (Fri, 29 Aug
2008) | 22 lines After working on the ao2_containers branch, I
noticed something a bit strange. In all cases where we provide a
callback function to ao2_container_alloc, the callback function
would only return 0 or CMP_MATCH. After inspecting the
ao2_callback() code carefully, I found that if you're only
looking for one specific item, then you should return CMP_MATCH |
CMP_STOP. Otherwise, astobj2 will continue traversing the current
bucket until the end searching for more matches. In cases like
chan_iax2 where in 1.4, all the peers are shoved into a single
bucket, this makes for potentially terrible performance since the
entire bucket will be traversed even if the peer is one of the
first ones come across in the bucket. All the changes I have made
were for cases where the callback function defined was passed to
ao2_container_alloc so that calls to ao2_find could find a unique
instance of whatever object was being stored in the container.
........
* main/file.c: Allow for video files to be opened as well as audio
files. (closes issue #13372) Reported by: epicac Patches:
13372.patch uploaded by putnopvut (license 60) Tested by: epicac
* /, apps/app_voicemail.c: Merged revisions 140421 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r140421 | mmichelson | 2008-08-29 11:01:07 -0500 (Fri,
29 Aug 2008) | 12 lines Add context checking when retrieving a
vm_state. This was causing a problem for people who had
identically named mailboxes in separate voicemail contexts. This
commit affects IMAP storage only. (closes issue #13194) Reported
by: moliveras Patches: 13194.patch uploaded by putnopvut (license
60) Tested by: putnopvut, moliveras ........
* channels/chan_sip.c: Merged revisions 140417 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r140417 | mmichelson | 2008-08-29 10:26:52 -0500 (Fri, 29 Aug
2008) | 10 lines Fix SIP's parsing so that if a port is specified
in a string to Dial(), it is not ignored. (closes issue #13355)
Reported by: acunningham Patches: 13355v2.patch uploaded by
putnopvut (license 60) Tested by: acunningham ........
2008-08-27 23:23 +0000 [r140355] Tilghman Lesher <tlesher@digium.com>
* cdr/cdr_pgsql.c: Oops
2008-08-27 20:11 +0000 [r140301] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Merged revisions 140299 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r140299 | mmichelson | 2008-08-27 14:49:20 -0500 (Wed, 27 Aug
2008) | 11 lines Fix tag checking in get_sip_pvt_byid_locked when
in pedantic mode. The problem was that the wrong tags would be
compared depending on the direction of the call. (closes issue
#13353) Reported by: flefoll Patches:
chan_sip.c.br14.139015.patch-refer-pedantic uploaded by flefoll
(license 244) ........
2008-08-26 21:59 +0000 [r140246] Doug Bailey <dbailey@digium.com>
* channels/chan_dahdi.c: Move the mwi send thread functionality
back into the do_monitor thread so that it is easier to manage
CID spill resources when do_monitor needs to be killed. (closes
issue #13213) Reported by: bbryant
2008-08-26 18:48 +0000 [r140205] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 140056 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r140056 | jpeeler | 2008-08-26 10:57:02 -0500 (Tue, 26
Aug 2008) | 9 lines (closes issue #12071) Reported by: tzafrir
Patches: dahdi_close.diff uploaded by tzafrir (license 46) Tested
by: tzafrir, jpeeler This patch fixes closing open file
descriptors in the case of an error. ........
2008-08-26 18:46 +0000 [r140201] Tilghman Lesher <tlesher@digium.com>
* apps/app_followme.c: OpenBSD compat fix (reminded by mvanbaak on
#asterisk-dev)
2008-08-26 18:11 +0000 [r140169] Russell Bryant <russell@digium.com>
* Makefile: Fix building menuselect-tree with PRINT_DIR set. We
_must_ use the --quiet flag here, or else some arbitrary text
will end up in the resulting menuselect-tree file and things will
explode.
2008-08-26 18:05 +0000 [r140167] Tilghman Lesher <tlesher@digium.com>
* configs/followme.conf.sample, apps/app_followme.c: Standardize
the option names for consistency (but continue to work with the
existing names for backwards compatibility). (closes issue
#13370) Reported by: jsturtevant
2008-08-26 16:10 +0000 [r140061] Russell Bryant <russell@digium.com>
* /, channels/chan_sip.c: Merged revisions 140060 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r140060 | russell | 2008-08-26 11:07:58 -0500 (Tue, 26 Aug 2008)
| 6 lines Fix some bogus scheduler usage in chan_sip. This code
used the return value of a completely unrelated function to
determine whether the scheduler should be run or not. This would
have caused the scheduler to not run in cases where it should
have. Also, leave a note about another scheduler issue that needs
to be addressed at some point. ........
2008-08-26 15:57 +0000 [r140057] Steve Murphy <murf@digium.com>
* main/cdr.c, configs/cdr.conf.sample, CHANGES,
include/asterisk/options.h: (closes issue #13366) Reported by:
erousseau This was a reasonable enhancement request, which was
easy to implement. Since it's an enhancement, it could only be
applied to trunk. Basically, for accounting where "initiated"
seconds are billed for, if the microseconds field on the end time
is greater than the microseconds field for the answer time, add
one second to the billsec field. The implementation was requested
by erousseau, and I've implemented it as requested. I've updated
the CHANGES, the cdr.conf.sample, and the .h files accordingly,
to accept and set a flag for the corresponding new option. cdr.c
adds in the extra second based on the usec fields if the option
is set. Tested, seems to be working fine.
2008-08-26 15:29 +0000 [r140053] Russell Bryant <russell@digium.com>
* /, channels/chan_iax2.c: Merged revisions 140051 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r140051 | russell | 2008-08-26 10:27:23 -0500 (Tue, 26
Aug 2008) | 15 lines Fix a race condition with the IAX scheduler
thread. A lock and condition are used here to allow newly
scheduled tasks to wake up the scheduler just in case the new
task needs to run sooner than the current wakeup time when the
thread is sleeping. However, there was a race condition such that
a newly scheduled task would not properly wake up the scheduler
or affect the wake up period. The order of execution would have
been: 1) Scheduler thread determines wake up time of N ms. 2)
Another thread schedules a task and signals the condition, with
an execution time of < N ms. 3) Scheduler thread locks and goes
to sleep for N ms. By moving the sleep time determination to
inside the critical section, this possibility is avoided.
........
2008-08-25 23:13 +0000 [r139981] Tilghman Lesher <tlesher@digium.com>
* Makefile, doc/asterisk.8, include/asterisk/options.h,
main/asterisk.c, main/term.c: Optional light colored background,
for those who use black on white terminals. (closes issue #13306)
Reported by: Corydon76 Patches: 20080814__bug13306__3.diff.txt
uploaded by Corydon76 (license 14) Tested by: Corydon76, pkempgen
2008-08-25 21:48 +0000 [r139928] Jeff Peeler <jpeeler@digium.com>
* main/manager.c, /: Merged revisions 139927 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r139927 | jpeeler | 2008-08-25 16:47:33 -0500 (Mon, 25 Aug 2008)
| 3 lines Fix a typo I made. Lesson learned, apply the patch if
one exists. ........
2008-08-25 21:32 +0000 [r139915] Sean Bright <sean.bright@gmail.com>
* build_tools/get_moduleinfo, /, build_tools/get_makeopts: Merged
revisions 139909 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r139909 | seanbright | 2008-08-25 17:31:03 -0400 (Mon, 25 Aug
2008) | 9 lines Some versions of awk (nawk, for example) don't
like empty regular expressions so be slightly more verbose.
(closes issue #13374) Reported by: dougm Patches: 13374.diff
uploaded by seanbright (license 71) Tested by: dougm ........
2008-08-25 20:59 +0000 [r139870] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c: Merged revisions 139869 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r139869 | twilson | 2008-08-25 15:46:10 -0500 (Mon, 25 Aug 2008)
| 2 lines Make SIPADDHEADER() propagate indefinitely ........
2008-08-25 17:24 +0000 [r139832] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Add output of variables to AgentRingNoAnswer
manager event if eventwhencalled is set to "vars" in queues.conf.
Yay for consistency. (closes issue #13369) Reported by: srt
Patches: 13369_agentringnoanswer_variables.diff uploaded by srt
(license 378)
2008-08-25 16:02 +0000 [r139775] Tilghman Lesher <tlesher@digium.com>
* doc/followme.txt (added), apps/app_followme.c: Realtime
capabilities for the Find-Me-Follow-Me application. (closes issue
#13295) Reported by: Corydon76 Patches:
20080813__followme_realtime_enabled.diff.txt uploaded by
Corydon76 (license 14) Tested by: dferrer
2008-08-25 15:54 +0000 [r139770] Steve Murphy <murf@digium.com>
* main/pbx.c, /, main/features.c: Merged revisions 139764 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r139764 | murf | 2008-08-25 09:33:14 -0600 (Mon, 25 Aug 2008) | 9
lines This patch reverts the changes made via 139347, and 139635,
as users are seeing adverse difference. I will un-close 13251.
Back to the drawing board/ concept/ beginning/ whatever! ........
2008-08-24 16:26 +0000 [r139704-139707] Tilghman Lesher <tlesher@digium.com>
* cdr/cdr_pgsql.c: Memory leak
* cdr/cdr_pgsql.c: Eliminate open coding of ast_str
2008-08-22 22:32 +0000 [r139627-139662] Steve Murphy <murf@digium.com>
* /, main/features.c: Merged revisions 139635 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r139635 | murf | 2008-08-22 16:24:02 -0600 (Fri, 22 Aug 2008) | 6
lines I found some problems with the code I committed earlier,
when I merged them into trunk, so I'm coming back to clean up.
And, in the process, I found an error in the code I added to
trunk and 1.6.x, that I'll fix using this patch also. ........
* apps/app_dial.c, main/pbx.c, /, main/features.c: Merged revisions
139347 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r139347 | murf | 2008-08-21 17:03:50 -0600 (Thu, 21 Aug 2008) |
47 lines (closes issue #13251) Reported by: sergee Tested by:
murf THis is a bold move for a static release fix, but I wouldn't
have made it if I didn't feel confident (at least a *bit*
confident) that it wouldn't mess everyone up. The reasoning goes
something like this: 1. We simply cannot do anything with CDR's
at the current point (in pbx.c, after the __ast_pbx_run loop).
It's way too late to have any affect on the CDRs. The CDR is
already posted and gone, and the remnants have been cleared. 2. I
was very much afraid that moving the running of the 'h' extension
down into the bridge code (where it would be now practical to do
it), would result in a lot more calls to the 'h' exten, so I
implemented it as another exten under another name, but found, to
my pleasant surprise, that there was a 1:1 correspondence to the
running of the 'h' exten in the pbx_run loop, and the new spot at
the end of the bridge. So, I ifdef'd out the current 'h' loop,
and moved it into the bridge code. The only difference I can see
is the stuff about the AST_PBX_KEEPALIVE, and hopefully, if this
is still an important decision point, I can replicate it if there
are complaints. To be perfectly honest, the KEEPALIVE situation
is not totally clear to me, and how it relates to a post-bridge
situation is less clear. I suspect the users will point out
everything in total clarity if this steps on anyone's toes! 3. I
temporarily swap the bridge_cdr into the channel before running
the 'h' exten, which makes it possible for users to edit the cdr
before it goes out the door. And, of course, with the
endbeforehexten config var set, the users can also get at the
billsec/duration vals. After the h exten finishes, the cdr is
swapped back and processing continues as normal. Please, all who
deal with CDR's, please test this version of Asterisk, and file
bug reports as appropriate! ........ I also made a little fix to
the app_dial's 'e' option, that is related to my updates.
2008-08-22 21:57 +0000 [r139622-139624] Jeff Peeler <jpeeler@digium.com>
* main/manager.c, /: Merged revisions 139621 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r139621 | jpeeler | 2008-08-22 16:36:13 -0500 (Fri, 22 Aug 2008)
| 5 lines (closes issue #13359) Reported by: Laureano Patches:
originate_channel_check.patch uploaded by Laureano (license 265)
........
* main/features.c: remove extra comma typo
2008-08-22 20:20 +0000 [r139457-139563] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: The -1 return value from incomplete or
improper headers for the SipNotify manager command was causing
the current manager session to become disconnected. Change the
return value to 0 for these cases. Also change a test for a NULL
pointer to be ast_strlen_zero instead. (closes issue #13351)
Reported by: Laureano Patches: sipnotify_action_fix.patch
uploaded by Laureano (license 265)
* main/features.c: Add missing unique id to ParkedCallGiveUp and
ParkedCallTimeOut manager events (closes issue #13358) Reported
by: srt Patches: 13358_parking_events.diff uploaded by srt
(license 378)
* /, include/asterisk/threadstorage.h: Merged revisions 139553 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r139553 | mmichelson | 2008-08-22 14:45:19 -0500 (Fri, 22 Aug
2008) | 8 lines Fix compilation when DEBUG_THREAD_LOCALS is
selected (closes issue #13298) Reported by: snuffy Patches:
bug13298_20080822.diff uploaded by snuffy (license 35) ........
* /, channels/chan_iax2.c: Merged revisions 139466 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r139466 | mmichelson | 2008-08-22 12:24:47 -0500 (Fri,
22 Aug 2008) | 3 lines Fix the build. Thanks, mvanbaak! ........
* /, channels/chan_iax2.c: Merged revisions 139456 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r139456 | mmichelson | 2008-08-22 11:57:38 -0500 (Fri,
22 Aug 2008) | 7 lines Prevent a deadlock in chan_iax2 resulting
from incorrect locking order between iax2_pvt and ast_channel
structures. AST-13 ........
2008-08-21 23:41 +0000 [r139391] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 139387 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r139387 | jpeeler | 2008-08-21 18:39:31 -0500 (Thu, 21
Aug 2008) | 3 lines Fixes loop that could possibly never exit in
the event of a channel never being able to be opened or specify
after a restart. (closes issue #11017) ........
2008-08-21 23:00 +0000 [r139345-139346] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
* apps/app_receivefax.c (removed), apps/app_sendfax.c (removed):
oops
* apps/app_receivefax.c (added), apps/app_sendfax.c (added):
initiate T38 negotiation in FaxSend; use channel variables; other
stuff too
2008-08-21 09:55 +0000 [r139281] Philippe Sultan <philippe.sultan@gmail.com>
* channels/chan_gtalk.c: Fix two memory leaks in chan_gtalk, thanks
Eliel! (closes issue #13310) Reported by: eliel Patches:
chan_gtalk.c.patch uploaded by eliel (license 64)
2008-08-20 22:16 +0000 [r139215] Russell Bryant <russell@digium.com>
* /, apps/app_chanspy.c: Merged revisions 139213 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r139213 | russell | 2008-08-20 17:14:35 -0500 (Wed, 20 Aug 2008)
| 11 lines Fix a crash in the ChanSpy application. The issue here
is that if you call ChanSpy and specify a spy group, and sit in
the application long enough looping through the channel list, you
will eventually run out of stack space and the application with
exit with a seg fault. The backtrace was always inside of a
harmless snprintf() call, so it was tricky to track down.
However, it turned out that the call to snprintf() was just the
biggest stack consumer in this code path, so it would always be
the first one to hit the boundary. (closes issue #13338) Reported
by: ruddy ........
2008-08-20 22:06 +0000 [r139210] Jason Parker <jparker@digium.com>
* channels/chan_sip.c: Fix output of sipshowpeer manager response.
(closes issue #13346) Reported by: srt Patches:
13346_malformed_sip_show_peer_response.diff uploaded by srt
(license 378)
2008-08-20 20:03 +0000 [r139153-139154] Shaun Ruffell <sruffell@digium.com>
* codecs/codec_dahdi.c: Remove extraneous debugging messages.
* codecs/codec_dahdi.c: Fix bug where the samples were not accurate
when in G723 mode, which would cause the timestamp field of the
RTP header to be invalid.
2008-08-20 17:25 +0000 [r139083] Steve Murphy <murf@digium.com>
* main/cdr.c, /: Merged revisions 139074 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r139074 | murf | 2008-08-20 11:14:55 -0600 (Wed, 20 Aug 2008) |
12 lines (closes issue #13263) Reported by: brainy Tested by:
murf The specialized reset routine is tromping on the flags field
of the CDR. I made a change to not reset the DISABLED bit. This
should get rid of this problem. ........
2008-08-20 16:16 +0000 [r139020] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_skinny.c: fix unholding phones after hangup on
older cisco phones. Patch by wedhorn.
2008-08-20 15:38 +0000 [r138887-139016] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 139015 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r139015 | mmichelson | 2008-08-20 10:37:56 -0500 (Wed, 20 Aug
2008) | 6 lines sip_read should properly handle a NULL return
from sip_rtp_read. (closes issue #13257) Reported by: travishein
........
* /, channels/chan_agent.c: Merged revisions 138942 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r138942 | mmichelson | 2008-08-19 18:17:17 -0500 (Tue,
19 Aug 2008) | 11 lines Reset agent_pvt variables back to the
values in agents.conf (from what the corresponding channel
variables were set to) when the agent logs out. (closes issue
#13098) Reported by: davidw Patches:
20080731__issue13098_agent_ackcall_not_reset.diff uploaded by
bbryant (license 36) Tested by: davidw ........
* /, apps/app_chanspy.c: Merged revisions 138886 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r138886 | mmichelson | 2008-08-19 13:50:53 -0500 (Tue, 19 Aug
2008) | 23 lines Add a lock and unlock prior to the destruction
of the chanspy_ds lock to ensure that no other threads still have
it locked. While this should not happen under normal
circumstances, it appears that if the spyer and spyee hang up at
nearly the same time, the following may occur. 1.
ast_channel_free is called on the spyee's channel. 2. The chanspy
datastore is removed from the spyee's channel in
ast_channel_free. 3. In the spyer's thread, the spyer attempts to
remove and destroy the datastore from the spyee channel, but the
datastore has already been removed in step 2, so the spyer
continues in the code. 4. The spyee's thread continues and calls
the datastore's destroy callback, chanspy_ds_destroy. This
involves locking the chanspy_ds. 5. Now the spyer attempts to
destroy the chanspy_ds lock. The problem is that in step 4, the
spyee has locked this lock, meaning that the spyer is attempting
to destroy a lock which is currently locked by another thread.
The backtrace provided in issue #12969 supports the idea that
this is possible (and has even occurred). This commit does not
close the issue, but should help in preventing one type of crash
associated with the use of app_chanspy. ........
2008-08-19 16:56 +0000 [r138851] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_skinny.c: chan_skinny now respects callwaiting=no
(closes issue #12691) Reported by: sbisker Patches:
callwaitingv1.diff uploaded by wedhorn (license 30) Tested by:
wedhorn on old skinny phones, mvanbaak on 7960 and 7905 with
latest firmware
2008-08-19 16:31 +0000 [r138815-138845] Steve Murphy <murf@digium.com>
* res/ael/ael.tab.c, res/ael/ael.y, res/ael/ael.tab.h,
utils/ael_main.c, utils/conf2ael.c: Oops. put a decl in a
generated file. My bad, but fixed now.
* main/pbx.c, res/ael/ael.tab.c, res/ael/ael.y, res/ael/ael.tab.h:
These changes are in regards to bug 13249, where users are being
surprised by the changes made to the Set app in trunk/1.6.x, as
they come from the 1.4 world. They are only bitten if they write
their AEL dialplan in the 1.4 world, and then carry it over to a
trunk/1.6.x installation where a "make samples" was executed, or
where they hand-edited the asterisk.conf file and added the
[compat] category with app_set = 1.6 (or higher). (this commit
does not totally solve 13249, at least not yet) The change
involves issueing a single warning while the AEL file is loading,
if: 1. app_set is present in the config file, and set to 1.6 or
higher. 2. there are double quotes in an assignment statement (eg
x = "hi there";) 3. the warning was not already issued. The
standalone app, aelparse, does not (yet) issue this warning. I'd
have to have it read in the asterisk.conf file, and that's a bit
of hassle. I'll add it if users request it, tho.
2008-08-19 15:58 +0000 [r138814] Philippe Sultan <philippe.sultan@gmail.com>
* res/res_jabber.c: Mention JID rather than SreenName in help
messages
2008-08-19 00:10 +0000 [r138775-138780] Sean Bright <sean.bright@gmail.com>
* channels/chan_sip.c: Let it compile now, too (woops)
* channels/chan_sip.c: And remove code we don't need anymore.
* channels/chan_sip.c: While we're at it, make this machine
parseable too.
* channels/chan_sip.c: Change event header to RegistrationTime to
be more consistent (and avoid breaking existing frameworks).
Pointed out by Laureano on #asterisk-dev.
2008-08-18 21:07 +0000 [r138738] Richard Mudgett <rmudgett@digium.com>
* channels/misdn/isdn_lib_intern.h, channels/misdn/isdn_lib.h,
doc/tex/misdn.tex, channels/chan_misdn.c,
configs/misdn.conf.sample, channels/misdn/isdn_lib.c,
channels/misdn_config.c: channels/chan_misdn.c * Made
bearer2str() use allowed_bearers_array[] * Made use the causes.h
defines instead of hardcoded numbers. * Made use Asterisk
presentation indicator values if either of the mISDN presentation
or screen options are negative. * Updated the misdn_set_opt
application option descriptions. * Renamed the awkward Caller ID
presentation misdn_set_opt application option value not_screened
to restricted. Deprecated the not_screened option value.
channels/misdn/isdn_lib.c * Made use the causes.h defines instead
of hardcoded numbers. * Fixed some spelling errors and typos. *
Added all defined facility code strings to fac2str().
channels/misdn/isdn_lib.h * Added doxygen comments to struct
misdn_bchannel. channels/misdn/isdn_lib_intern.h * Added doxygen
comments to struct misdn_stack. channels/misdn_config.c
configs/misdn.conf.sample * Updated the mISDN presentation and
screen parameter descriptions. doc/tex/misdn.tex * Updated the
misdn_set_opt application option descriptions. * Fixed some
spelling errors and typos.
2008-08-18 20:23 +0000 [r138687-138694] Mark Michelson <mmichelson@digium.com>
* configs/queues.conf.sample, apps/app_queue.c: Change the queue
timeout priority logic into less ugly and confusing code pieces.
Clarify the logic within queues.conf.sample. (closes issue
#12690) Reported by: atis Patches: queue_timeoutpriority.patch
uploaded by atis (license 242)
* apps/app_queue.c: Merged revisions 138685 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r138685 | mmichelson | 2008-08-18 15:01:14 -0500 (Mon, 18 Aug
2008) | 10 lines Change the inequalities used in app_queue with
regards to timeouts from being strict to non-strict for more
accuracy. (closes issue #13239) Reported by: atis Patches:
app_queue_timeouts_v2.patch uploaded by atis (license 242)
........
2008-08-18 15:54 +0000 [r138631] Jason Parker <jparker@digium.com>
* Makefile: Remove option that isn't valid here.
2008-08-18 02:13 +0000 [r138518] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c: add missing define for SS7 in
dahdi_restart
2008-08-17 14:12 +0000 [r138442-138482] Sean Bright <sean.bright@gmail.com>
* main/features.c: Move Uniqueid to the end of the event for those
that rely on the position of the name/value pairs, pointed out by
snuffy-home on #asterisk-commits. For those of you who rely on
the position of name/value pairs in manager events... stop...
that is why associative arrays were invented.
* main/features.c: Add Uniqueid header to ParkedCall manager event.
(closes issue #13323) Reported by: srt Patches:
13323_unique_id_for_parkedcalls_event.diff uploaded by srt
(license 378)
* main/rtp.c: Add missing colons to RTCPReceived and RTCPSent
manager events. (closes issue #13319) Reported by: srt Patches:
13319_rtcp_manager_event_headers.diff uploaded by srt (license
378)
* channels/chan_iax2.c: Fix the output of the JitterBufStats
manager event. (closes issue #13324) Reported by: srt Patches:
13324_missing_nl_in_jitterbufstats_event_2.diff uploaded by srt
(license 378)
* configs/cdr_tds.conf.sample: Since it's introduction in revision
3497, cdr_tds has *never* read the port configuration option from
cdr_tds.conf. So go ahead and remove it from the sample config.
2008-08-16 13:07 +0000 [r138409-138412] Tilghman Lesher <tlesher@digium.com>
* channels/chan_dahdi.c: Fix compilation warnings (found with
dev-mode)
* main/pbx.c: Also make sure hinting won't crash on reload. (Closes
issue #13312)
2008-08-16 01:13 +0000 [r138311-138361] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 138360 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r138360 | jpeeler | 2008-08-15 20:12:18 -0500 (Fri, 15
Aug 2008) | 1 line fixes use count to properly decrement if an
active dahdi channel is destroyed allowing module to be unloaded
........
* channels/chan_dahdi.c, /: Merged revisions 138119,138151,138238
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r138119 | jpeeler | 2008-08-15 14:21:51 -0500 (Fri, 15 Aug 2008)
| 4 lines Fixes the dahdi restart functionality. Dahdi restart
allows one to restart all DAHDI channels, even if they are
currently in use. This is different from unloading and then
loading the module since unloading requires the use count to be
zero. Reloading the module is different in that the signalling is
not changed from what it was originally configured. Also, this
fixes not closing all the file descriptors for D-channels upon
module unload (which would prevent loading the module
afterwards). (closes issue #11017) ........ r138151 | jpeeler |
2008-08-15 14:41:29 -0500 (Fri, 15 Aug 2008) | 1 line declared
static mutexes using AST_MUTEX_DEFINE_STATIC macro ........
r138238 | jpeeler | 2008-08-15 16:28:26 -0500 (Fri, 15 Aug 2008)
| 1 line initialize condition variable ss_thread_complete using
ast_cond_init ........
2008-08-15 22:54 +0000 [r138206-138260] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
138258 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r138258 | tilghman | 2008-08-15 17:33:42 -0500 (Fri, 15 Aug 2008)
| 8 lines More fixes for realtime peers. (closes issue #12921)
Reported by: Nuitari Patches: 20080804__bug12921.diff.txt
uploaded by Corydon76 (license 14) 20080815__bug12921.diff.txt
uploaded by Corydon76 (license 14) Tested by: Corydon76 ........
* main/pbx.c, configs/extensions.conf.sample: Remove deprecated
syntax from sample config file (closes issue #13314) Reported by:
kue
2008-08-15 20:12 +0000 [r138155] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c: rename all zfd instances in chan_dahdi to
dfd to match 1.4 (left over from DAHDI transition)
2008-08-15 19:36 +0000 [r138086-138148] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c: Change free to ast_free_ptr, too
* main/pbx.c: e->data can be NULL, so use the safe version of
ast_strdup() (closes issue #13312) Reported by: pj
* channels/chan_sip.c: regseconds is actually stored as the epoch
time, not registration length
2008-08-15 15:09 +0000 [r138028] Russell Bryant <russell@digium.com>
* main/autoservice.c, /: Merged revisions 138027 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r138027 | russell | 2008-08-15 10:07:16 -0500 (Fri, 15 Aug 2008)
| 9 lines Ensure that when a hangup occurs in autoservice, that a
hangup frame gets properly deferred to be read from the channel
owner when it gets taken out of autoservice. (closes issue
#12874) Reported by: dimas Patches: v1-12874.patch uploaded by
dimas (license 88) ........
2008-08-15 15:03 +0000 [r138024] Tilghman Lesher <tlesher@digium.com>
* /, funcs/func_strings.c: Merged revisions 138023 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r138023 | tilghman | 2008-08-15 09:51:12 -0500 (Fri, 15
Aug 2008) | 8 lines Additional check for more string specifiers
than arguments. (closes issue #13299) Reported by: adomjan
Patches: 20080813__bug13299.diff.txt uploaded by Corydon76
(license 14) func_strings.c-sprintf.patch uploaded by adomjan
(license 487) Tested by: adomjan ........
2008-08-14 22:43 +0000 [r137987] Russell Bryant <russell@digium.com>
* doc/tex/Makefile: Fix a bashism that causes an error when trying
to build the pdf on ubuntu
2008-08-14 18:47 +0000 [r137933] Sean Bright <sean.bright@gmail.com>
* cdr/cdr_sqlite3_custom.c: Fix memory leak in cdr_sqlite3_custom.
(closes issue #13304) Reported by: eliel Patches: sqlite.patch
uploaded by eliel (license 64) (Slightly modified by me)
2008-08-14 18:12 +0000 [r137901] Russell Bryant <russell@digium.com>
* CHANGES: Prepare for adding 1.6.2 changes
2008-08-14 16:52 +0000 [r137848] Tilghman Lesher <tlesher@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 137847 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r137847 | tilghman | 2008-08-14 11:47:30 -0500 (Thu, 14
Aug 2008) | 9 lines When creating the secondary subchannel name,
it is necessary to compare to the existing channel name without
the "Zap/" or "DAHDI/" prefix, since our test string is also
without that prefix. (closes issue #13027) Reported by: dferrer
Patches: chan_zap-1.4.21.1_fix2.patch uploaded by dferrer
(license 525) (Slightly modified by me, to compensate for both
names) ........
2008-08-14 15:32 +0000 [r137812] Jason Parker <jparker@digium.com>
* channels/chan_sip.c: Make sure we set the socket port, so we
don't try to use <ip address>:0. (closes issue #13255) Reported
by: falves11 Patches: 13255-socketport.diff uploaded by qwell
(license 4) Tested by: falves11
2008-08-14 15:03 +0000 [r137780] Sean Bright <sean.bright@gmail.com>
* cdr/cdr_tds.c: If we detect that we are no longer connected, try
to reconnect a few times before giving up. This relies on the
timeout settings in the freetds.conf file and, unfortunately, on
a recent version of FreeTDS (0.82 or newer). I either need to
change the current execs to be non-blocking (which I do not want
to do) or we have to force people to run with the latest and
greatest of FreeTDS. I'm on the fence...
2008-08-14 14:15 +0000 [r137732] Russell Bryant <russell@digium.com>
* /, configs/sip.conf.sample: Merged revisions 137731 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r137731 | russell | 2008-08-14 09:05:23 -0500 (Thu, 14
Aug 2008) | 4 lines Comments in this config file were aligned
only if your tab size was set to 8. So, convert tabs to spaces so
that things should be aligned regardless of what tab size you use
in your editor. ........
2008-08-14 02:03 +0000 [r137680] Kevin P. Fleming <kpfleming@digium.com>
* /, Zaptel-to-DAHDI.txt: Merged revisions 137679 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r137679 | kpfleming | 2008-08-13 21:03:04 -0500 (Wed, 13 Aug
2008) | 1 line forgot one module name that changed ........
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