https://origsvn.digium.com/svn/asterisk/branches/10
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r329200 | rmudgett | 2011-07-21 12:32:02 -0500 (Thu, 21 Jul 2011) | 24 lines
Merged revisions 329199 via svnmerge from
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r329199 | rmudgett | 2011-07-21 12:30:57 -0500 (Thu, 21 Jul 2011) | 17 lines
Update PickupChan documentation.
The PickupChan uses the ampersand as the argument separator.
Was documented as:
PickupChan(channel[,channel2[,...][,options]])
Fixed documentation to:
PickupChan(Technology/Resource[&Technology2/Resource2[&...]][,options])
This is a continuation of ASTERISK-17494 for v1.8 and later.
(closes issue ASTERISK-18144)
Reported by: Erik Smith
Patches:
pickupchan_ducumentation-v2.patch (License #6263) patch uploaded by Erik Smith
Tested by: Erik Smith
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r328992 | twilson | 2011-07-20 15:18:25 -0500 (Wed, 20 Jul 2011) | 12 lines
Merged revisions 328987 via svnmerge from
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r328987 | twilson | 2011-07-20 15:16:58 -0500 (Wed, 20 Jul 2011) | 5 lines
We can't guarantee an eth0 is present
FreeBSD test fails on this case presumably because there is no eth0 on the test
machine. Better to just remove this test for now.
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r328936 | kmoore | 2011-07-20 14:01:37 -0500 (Wed, 20 Jul 2011) | 15 lines
Merged revisions 328935 via svnmerge from
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r328935 | kmoore | 2011-07-20 14:00:23 -0500 (Wed, 20 Jul 2011) | 8 lines
Inband DTMF regression
The functionality of inband DTMF in chan_sip relied upon
ast_rtp_instance_dtmf_mode_get/set not working properly to avoid calling
ast_rtp_instance_dtmf_begin/end on RTP streams with inband DTMF. According to
documentation, ast_rtp_instance_dtmf_begin/end is meant only for RFC2833 DTMF,
never inband. This fixes the regression introduced in revision 328823.
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r328879 | kpfleming | 2011-07-19 16:31:16 -0500 (Tue, 19 Jul 2011) | 23 lines
Merged revisions 328878 via svnmerge from
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r328878 | kpfleming | 2011-07-19 16:29:07 -0500 (Tue, 19 Jul 2011) | 17 lines
Revert partial attempt at handling pathnames with spaces.
Revision 299794 attempted to improve the build system to be able to handle
pathnames (primarily DESTDIR) with spaces in them, since this is common on
some platforms (including Mac OSX). Unfortunately, the changes were incomplete
and did not actually provide the desired behavior, and as a side effect the
functionality that ensured that stale headers in the Asterisk 'include' directory
were removed got broken. In addition, the check for stale (and possibly
incompatible) modules in the Asterisk 'modules' directory also got broken, and
would never report any stale modules. Users upgrading to this version or later
versions would then see unexpected module load errors.
Since there are few users who actually want to install Asterisk into paths
that contain spaces, and a proper fix for the build system would take many hours,
the best solution for now is to just revert the partial solution.
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r328824 | kmoore | 2011-07-19 13:05:21 -0500 (Tue, 19 Jul 2011) | 18 lines
Merged revisions 328823 via svnmerge from
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r328823 | kmoore | 2011-07-19 12:57:18 -0500 (Tue, 19 Jul 2011) | 11 lines
RTP bridge away with inband DTMF and feature detection
When deciding whether Asterisk was allowed to bridge the call away from the
core, chan_sip did not take into account the usage of features on dialed
channels that require monitoring of DTMF on channels utilizing inband DTMF.
This would cause Asterisk to allow the call to be locally or remotely bridged,
preventing access to the data required to detect activations of such features.
(closes 17237)
Review: https://reviewboard.asterisk.org/r/1302/
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r328771 | kmoore | 2011-07-19 10:46:54 -0500 (Tue, 19 Jul 2011) | 18 lines
Merged revisions 328770 via svnmerge from
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r328770 | kmoore | 2011-07-19 10:43:32 -0500 (Tue, 19 Jul 2011) | 11 lines
MeetMe requests a PIN twice in some circumstances
If a call to MeetMe includes both the dynamic(D) and always request PIN(P)
options, MeetMe will ask for the PIN two times: once for creating the
conference and once for entering the conference. This behavior was introduced
in rev 311616 when adding the CONFFLAG_ALWAYSPROMPT option to the logic branch
controlling PIN entry for joining a conference.
(closes AST-601)
Review: https://reviewboard.asterisk.org/r/1305/
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r328664 | markm | 2011-07-18 16:50:13 -0400 (Mon, 18 Jul 2011) | 15 lines
Merged revisions 328663 via svnmerge from
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r328663 | markm | 2011-07-18 16:47:04 -0400 (Mon, 18 Jul 2011) | 9 lines
app_dial may double free a channel datastore
When starting a call with originate, and having the callee channel run Bridge() on pickup, we will double free the dialed_interface_info datastore, causing a crash. Make sure to check if the datastore still exists before trying to free it.
(closes issue ASTERISK-17917)
Reported by: Mark Murawski
Tested by: Mark Murawski
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r328611 | markm | 2011-07-18 08:56:49 -0400 (Mon, 18 Jul 2011) | 15 lines
Merged revisions 328608 via svnmerge from
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r328608 | markm | 2011-07-18 08:35:57 -0400 (Mon, 18 Jul 2011) | 9 lines
If the sip private structure is null, sip_setoption() will defref the null pointer and crash.
Ideally, sip_setoption shouldn't be called if there is a lack of a sip private structure. But this will fix a crash.
(closes issue ASTERISK-17909)
Reported by: Mark Murawski
Tested by: Mark Murawski
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r328120 | dvossel | 2011-07-13 17:09:34 -0500 (Wed, 13 Jul 2011) | 15 lines
Preserve sample rate quality of wideband mixmonitor recordings.
MixMonitor has the ability to record in any file format Asterisk supports,
but the quality of wideband audio is not preserved. This is because
regardless of the sample rate the call is being recorded in, the audio
is always downsampled to 8khz and then upsampled to whatever wideband
format it is being written as. This patch resolves this by requesting
the audio from the audiohook in the signed linear format closest to the
sample rate of the format we are writing. This fix is only possible for
Asterisk 1.10 because audio hooks in 1.8 are not capable of wideband
audio.
Review: https://reviewboard.asterisk.org/r/1314/
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r327950 | kpfleming | 2011-07-12 17:53:53 -0500 (Tue, 12 Jul 2011) | 14 lines
Correct double-free situation in manager output processing.
The process_output() function calls ast_str_append() and xml_translate() on its
'out' parameter, which is a pointer to an ast_str buffer. If either of these
functions need to reallocate the ast_str so it will have more space, they will
free the existing buffer and allocate a new one, returning the address of the
new one. However, because process_output only receives a pointer to the ast_str,
not a pointer to its caller's variable holding the pointer, if the original
ast_str is freed, the caller will not know, and will continue to use it (and
later attempt to free it).
(reported by jkroon on #asterisk-dev)
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r327852 | mjordan | 2011-07-12 14:10:34 -0500 (Tue, 12 Jul 2011) | 12 lines
Added additional checks for mailbox / password beginning with '*' character
A bug existed such that if a user entered a password with '*', and the extension 'a' did not exist, an invalid mailbox would be created and the user authenticated. The code was changed to prevent this from occurring, and to prevent users from having mailboxes or passwords defined that begin with the '*' character.
(closes issue ASTERISK-17443)
Reported by: Kevin Scott Adams
Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1316/
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r327793 | tilghman | 2011-07-12 10:35:46 -0500 (Tue, 12 Jul 2011) | 14 lines
Use 'printf' (POSIX issue 4) instead of 'echo -n', for portability.
The problem with using 'echo -n' is that it is not portable. While BSD systems
required that the '-n' option be removed and interpreted, System V required
that all strings should be echoed with no interpretation of options. This
fundamental difference of behavior means that it is never possible to use the
'-n' flag to echo in tests which are meant to be portable.
In this case, on Mac OS X 10.6, the /bin/sh shell builtin 'echo' uses the
System V semantics of the command, and thus the SHELL test failed on that
platform.
http://pubs.opengroup.org/onlinepubs/009695399/utilities/echo.html#tag_04_41_16
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When undergoing a shutdown and channels are kicked out of a bridge, a segfault
occurs because ConfBridge tries to play sounds on the bridge after the
underlying channels have been blown away due to the shutdown.
(closes ASTERISK-18040)
Review: https://reviewboard.asterisk.org/r/1283/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@327748 f38db490-d61c-443f-a65b-d21fe96a405b
follow_talker mode originally echoed the same video stream
to all participants. As the primary talker switched around, the
video stream would result in the talker seeing themselves. Now
the primary talker sees the last person who was talking rather than
themselves.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@327640 f38db490-d61c-443f-a65b-d21fe96a405b
generating party time to send its own T.38 reinvite.
Also don't forward frames through the gateway if we are negotiating T.38.
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r327211 | rmudgett | 2011-07-08 16:41:58 -0500 (Fri, 08 Jul 2011) | 9 lines
INVITE 403 Forbidden response always retransmits the maximum times.
Asterisk sends a 403 Forbidden response if authentication fails for an
INVITE as required. However, it ignores the ACK and keeps retransmitting
the response.
* Made not delete the to-tag in the dialog so the expected ACK can be
matched with the dialog and stop the retransmissions.
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It was inconsistent to have the silk and celt format attribute
modules in the format file interpreter folder.
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r327106 | mnicholson | 2011-07-08 14:52:51 -0500 (Fri, 08 Jul 2011) | 11 lines
Reset our ast_str before passing it on to dialplan function backends.
It is possible for a dialplan backend to not modify the given buffer or ast_str
and still return success. This causes any previous value stored in the buffer
to be used as if the new function call provided it. Some functions also append
to the given buffer assuming it is empty.
The test_substitution unit test has also been modified to detect this problem.
(closes issue ASTERISK-17878)
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r326985 | rmudgett | 2011-07-07 20:08:05 -0500 (Thu, 07 Jul 2011) | 12 lines
Some code cleanup in pbx.c
* Mostly comment and format changes.
* ast_context_remove_extension_callerid() and ast_add_extension_nolock()
will write lock the found specific context.
* ast_context_find() will now tolerate a NULL name.
* Eliminated some inlined versions of find_context() and
find_context_locked().
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This patch adds pass-through support for CELT. CELT
formats are defined in codecs.conf and can be configured
to any sample rate a CELT endpoint supports. This patch also
addresses a crash in channel.c resulting from a frame list being
freed incorrectly. This crash was discovered while testing a CELT
translator which had to split encoded audio into multiple frames.
The codec translator is not a part of this patch, but may be
contributed in the future.
Review: https://reviewboard.asterisk.org/r/1294/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@326855 f38db490-d61c-443f-a65b-d21fe96a405b
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r326689 | jrose | 2011-07-07 11:04:51 -0500 (Thu, 07 Jul 2011) | 10 lines
res_odbc patch by tilghman to fix integers with null values
Addresses some improper sql statements in res_odbc that would cause an update to fail on
realtime peers due to trying to set as "(NULL)" rather than an actual NULL.
(closes issue #1922STERISK-17791)
Reported by: marcelloceschia
Patches:
20110505__issue19223.diff.txt uploaded by tilghman (license 14)
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Adds a new sub-command, "valgrind" to live_ast. It runs asterisk under
valgrind. The extra command-line parameters are passed to Asterisk as
usual, and parameters to valgrind are passed through LIVE_AST_VALGRIND_ARGS
in live.conf .
Review: https://reviewboard.asterisk.org/r/1109/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@326636 f38db490-d61c-443f-a65b-d21fe96a405b
There were some bugs in the very ancient version of Berkeley DB that Asterisk
used. Instead of spending the time tracking down the bugs in the Berkeley code
we move to the much better documented SQLite 3.
Conversion of the old astdb happens at runtime by running the included
astdb2sqlite3 utility. The ast_db API with SQLite 3 backend should behave
identically to the old Berkeley backend, but in the future we could offer a
much more robust interface.
We do not include the SQLite 3 library in the source tree, but instead rely
upon the distribution-provided libraries. SQLite is so ubiquitous that this
should not place undue burden on administrators.
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r326484 | dvossel | 2011-07-06 10:26:49 -0500 (Wed, 06 Jul 2011) | 10 lines
Reverts fix for timerfd locking issue.
jrose discovered a performance issue with this
fix that prevents his analog phones from working
when using timerfd as a timing source. Until
it is understood what is causing this performance
problem, this patch is being reverted.
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r326411 | tilghman | 2011-07-05 17:08:29 -0500 (Tue, 05 Jul 2011) | 14 lines
Add the attribute "type" to each "<use>" for menuselect.
This matters only when autoconf fails to detect that weak linking is supported.
External optional dependencies will become optional in both cases, as they are
removed at compile time when not detected. However, runtime-optional modules
are made mandatory when weak linking is not found. This change affects only
the external optional dependencies; previously, they were incorrectly required
when weak linking support was not detected.
Patches:
20110702__issue18062__asterisk_trunk.diff.txt by tilghman (License #5003)
Tested by: iasgoscouk
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Several variables in the script control which files are converted and the
source and destination formats.
Patch-by: Trey Blancher <support@digium.com>
(closes AST-560)
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r326291 | rmudgett | 2011-07-05 12:22:59 -0500 (Tue, 05 Jul 2011) | 23 lines
Used auth= parameter freed during "sip reload" causes crash.
If you use the auth= parameter and do a "sip reload" while there is an
ongoing call. The peer->auth data points to free'd memory.
The patch does several things:
1) Puts the authentication list into an ao2 object for reference counting
to fix the reported crash during a SIP reload.
2) Converts the authentication list from open coding to AST list macros.
3) Adds display of the global authentication list in "sip show settings".
(closes issue ASTERISK-17939)
Reported by: wdoekes
Patches:
jira_asterisk_17939_v1.8.patch (license #5621) patch uploaded by rmudgett
Review: https://reviewboard.asterisk.org/r/1303/
JIRA SWP-3526
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This adds a new action, FilterAdd to the manager interface that allows control over event filters for the current session
(closes issue ASTERISK-16795)
Reported by: kobaz
Tested by: kobaz,loloski
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r326209 | mjordan | 2011-07-05 08:23:57 -0500 (Tue, 05 Jul 2011) | 7 lines
Updated filestream destructor to block until move is complete when cache is used
When a cache directory is used, the process is forked and a mv command is executed to move the temporary file to the permanent location. This caused issues with voicemail, where a race condition occurred when the parent expected the file to be in the permanent location prior to the mv command completing. The parent process is now blocked until the mv command completes.
(closes issue ASTERISK-17724)
Reported by: Adiren P.
Tested by: mjordan
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r326144 | rmudgett | 2011-07-01 16:07:22 -0500 (Fri, 01 Jul 2011) | 16 lines
Better way to get chan and pvt lock for issue ASTERISK-17431.
Redoes -r308945 for issue ASTERISK-17431 deadlock fix for
sip_set_udptl_peer() and sip_set_rtp_peer().
* Lock the channels in the defined order and avoid the need for a deadlock
avoidance loop.
* Lock the channel before getting the pointer to the private structure to
be sure that the pointer will not change due to a masquerade or channel
hangup.
* To preserve sanity, check that chan and p->owner are the same. (Pointer
rearangements should not happen without the protection of locks because
bad things tend to happen otherwise.)
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r325935 | rmudgett | 2011-06-30 15:39:45 -0500 (Thu, 30 Jun 2011) | 11 lines
Misc minor changes in chan_sip.
* Add load failure exit if primary SIP container(s) could not get created
in chan_sip.c:load_module().
* Removed a redundant static prototype.
* Some typos.
* Some whitespace.
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r325821 | jrose | 2011-06-30 14:17:32 -0500 (Thu, 30 Jun 2011) | 10 lines
Fixes an issue with Music on Hold classes losing files in playlist when realtime is used.
The bug occurs rather intermittently and I relied on the reporters to test the patch.
After a sanity check and some testing, I'm giving it an OK.
(closes issue ASTERISK-17875)
Reported by: David Cunningham
Patches:
res_musiconhold.c.mohrt17875_v1 uploaded by Igor Goncharovsky (license #5009)
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terminal). Can be enabled on a channel by setting FAXOPT(gateway)=yes in the
dialplan.
Big thanks to irroot for porting this code to use the framehooks api.
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r325610 | rmudgett | 2011-06-29 13:05:15 -0500 (Wed, 29 Jun 2011) | 18 lines
Response to QueueRule manager command does not contain ActionID if it was specified.
* Add ActionID support as documented for the QueueRule AMI action.
* Remove documentation for ActionID with the Queues AMI action. The
output does not follow normal AMI response output and there is no place to
put an ActionID header.
(closes issue AST-602)
Reported by: Vlad Povorozniuc
Patches:
jira_ast_602_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: Vlad Povorozniuc, rmudgett
Review: https://reviewboard.asterisk.org/r/1295/
JIRA SWP-3575
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* Added general option negative_penalty_invalid default off. when set
members are seen as invalid/logged out when there penalty is negative.
for realtime members when set remove from queue will set penalty to -1.
* Added queue option autopausedelay when autopause is enabled it will be
delayed for this number of seconds since last successful call if there
was no prior call the agent will be autopaused immediately.
* Added member option ignorebusy this when set and ringinuse is not
will allow per member control of multiple calls as ringinuse does for
the Queue.
- Mark QUEUE_MEMBER_PENALTY Deprecated it never worked for realtime members
- QUEUE_MEMBER is now R/W supporting setting paused, ignorebusy and penalty.
(closes issue ASTERISK-17421)
(closes issue ASTERISK-17391)
Reported by: irroot
Tested by: irroot, jrose
Review: https://reviewboard.asterisk.org/r/1119/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@325483 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r325212 | rmudgett | 2011-06-28 12:30:16 -0500 (Tue, 28 Jun 2011) | 7 lines
Use the device name and not the channel name to initialize the device state.
Correct ASTERISK-11323 implementation as I don't see how it ever worked as
claimed when it used the channel name and not the device name.
(issue ASTERISK-11323)
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If a SDP does not modify the session, we ignore it. However, we were defaulting
no text and video support to true before checking to see if the sdp modified
anything or not. This would result in process_sdp ignoring an sdp but removing
video and text from the call during direct media reinvites.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@325151 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r324955 | tilghman | 2011-06-27 11:30:50 -0500 (Mon, 27 Jun 2011) | 5 lines
Save and restore errno from within signal handlers.
This is recommended by the POSIX standard, as well as by the sigaction(2) manpage
for various platforms that we support (e.g. Mac OS X).
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r324914 | rmudgett | 2011-06-27 10:37:19 -0500 (Mon, 27 Jun 2011) | 21 lines
When subscribing MWI to an unsolicited mailbox the first notification is incorrect.
A remote peer subscribed to MWI with the unsolicited option and a local
phone subscribed to the remote mailbox. The notify message-summary events
are sent correctly except for the first one when subscribing, which will
always be 0. This means the phone MWI indicator will be wrong until the
mailbox read/unread count changes and the event is fired.
Looks like this is a regression from ASTERISK-16149.
* Fix the logic to check the cache and if allowed then fallback to
manually counting mailbox messages.
(closes issue ASTERISK-17997)
Reported by: rsw686
Patches:
jira_asterisk_17997_v1.8.patch (license #5621) uploaded by rmudgett
Tested by: rsw686
JIRA SWP-3551
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r324849 | rmudgett | 2011-06-24 15:46:01 -0500 (Fri, 24 Jun 2011) | 15 lines
Syntax errors in dialplan do not display the file name.
When issuing the CLI command "dialplan reload" syntax errors and warnings
are displayed on the console. The offending line number is displayed on
the console, but the file name is not displayed. Errors caught in
main/config.c do display the file name.
(closes issue ASTERISK-17985)
Reported by: ulogic
Patches:
pbx_config.patch uploaded by ulogic (License #5685) modified format
Tested by: rmudgett
JIRA SWP-3554
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r324768 | jrose | 2011-06-24 11:48:06 -0500 (Fri, 24 Jun 2011) | 11 lines
DTMF wasn't being logged on connected consoles when enabled in logger.conf
Previously in order for DTMF to be logged in a connected console session, the user would
have to do logger set channel DTMF on. This corrects that so that it is on by default.
This issue was caused by an off by one error incurred by a logger level count of 6 in
logger.h where it should have been 7.
(closes issue: ASTERISK-17974)
Reported by: Luke H
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r324557 | twilson | 2011-06-22 22:10:38 -0500 (Wed, 22 Jun 2011) | 5 lines
Remove tests for parsing address with invalid port
getaddrinfo on OS X returns with EAI_NONAME error when passed a port
greater than 65535. Linux throws no error, so remove the tests for now.
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r324484 | twilson | 2011-06-22 13:52:04 -0500 (Wed, 22 Jun 2011) | 20 lines
Stop sending IPv6 link-local scope-ids in SIP messages
The idea behind the patch listed below was used, but in a more targeted manner.
There are now address stringification functions for addresses that are meant to
be sent to a remote party. Link-local scope-ids only make sense on the machine
from which they originate and so are stripped in the new functions.
There is also a host sanitization function added to chan_sip which is used
for when peer and dialog tohost fields or sip_registry hostnames are used to
craft a SIP message.
Also added are some basic unit tests for netsock2 address parsing.
(closes issue ASTERISK-17711)
Reported by: ch_djalel
Patches:
asterisk-1.8.3.2-ipv6_ll_scope.patch uploaded by ch_djalel (license 1251)
Review: https://reviewboard.asterisk.org/r/1278/
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Also fixed a reference leak in an error path in sip_msg_send().
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r324481 | rmudgett | 2011-06-22 13:41:20 -0500 (Wed, 22 Jun 2011) | 19 lines
Timout or error on INFO or MESSAGE transaction causes call to be lost.
When exchanging INFO messages within a call, 4xx error causes the call to
be disconnected although RFC 2976 explicitly states that such transactions
do not modify the state of the dialog.
When exchanging MESSAGE messages within a call, 4xx error causes the call
to be disconnected. To provide least surprise, we should not disconnect
the call since a MESSAGE is like INFO in this case. (Implied by RFC 3428
Section 2)
(closes issue ASTERISK-17901)
Reported by: neutrino88
Review: https://reviewboard.asterisk.org/r/1257/
Review: https://reviewboard.asterisk.org/r/1258/
JIRA SWP-3486
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r324364 | dvossel | 2011-06-21 15:11:52 -0500 (Tue, 21 Jun 2011) | 10 lines
Fixes locking inversion issue in ast_async_goto()
During this function we can not hold the "chan" lock while
doing the masquerade, the explicit goto on the tmp chan, or
the channel alloc. Instead we need to get the channel lock,
store off information about the channel that we need, and
then let the channel lock go for the remainder of the function.
Review: https://reviewboard.asterisk.org/r/1275/
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When playing back a prompt to a channel, confbridge neglects to check for
hangup events causing lockup condititions for hangups that occur before
actually joining the conference. This change ensures that the user is removed
from the conference in the event of a premature hangup.
Review: https://reviewboard.asterisk.org/r/1277/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@324304 f38db490-d61c-443f-a65b-d21fe96a405b
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r324237 | twilson | 2011-06-20 12:33:07 -0500 (Mon, 20 Jun 2011) | 12 lines
Ignore media offers with a port of 0
Section 5.1 of RFC3264 states:
A port number of zero in the offer indicates that the stream is offered
but MUST NOT be used.
(closes issue ASTERISK-17845)
Reported by: jacco
Patches:
issue19281_2.patch uploaded by jacco (license 1277)
Tested by: jacco, twilson
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r323754 | twilson | 2011-06-15 13:21:52 -0500 (Wed, 15 Jun 2011) | 23 lines
Merged revisions 323733 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r323733 | twilson | 2011-06-15 13:13:00 -0500 (Wed, 15 Jun 2011) | 16 lines
Merged revisions 323732 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r323732 | twilson | 2011-06-15 13:06:24 -0500 (Wed, 15 Jun 2011) | 9 lines
Fix DYNAMIC_FEATURES
DYNAMIC_FEATURES were broken by a recent DTMF change. This patch makes
sure that dynamic features are also checked when deciding whether or not
to pass DTMF through or store it for interpreting.
(closes issue ASTERISK-17914)
Reported by: vrban
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r323730 | jrose | 2011-06-15 12:42:42 -0500 (Wed, 15 Jun 2011) | 11 lines
Adds locking to find_table in res_configure_pgsql to prevent a crash.
Bryonclark described the problem as occuring during this function because of multiple
simultaneous database operations causing corruption against a pgsqlConn object.
(closes issue ASTERISK-17811)
Reported by: byronclark
Patches:
pgsql_find_table_locking.patch uploaded by byronclark (license 1200)
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r323669 | rmudgett | 2011-06-15 11:43:18 -0500 (Wed, 15 Jun 2011) | 21 lines
[regression] Voicemail MWI is no longer sent.
When leaving a voicemail, the MWI message is never sent. The same thing
happens when checking a voicemail and marking it as read.
If you restart Asterisk, everything comes up at that state correctly, but
changes to the messages in voicemail causes the light to not be set
appropriately. Very easy to reproduce.
* Made ast_event_check_subscriber() return TRUE if there are ANY
subscribers to an event type when there are no restricting ie values
passed. This allows an event being queued to be queued.
(closes issue ASTERISK-18002)
Reported by: lmadsen
Tested by: lmadsen, irroot
Patches:
jira_asterisk_18002_v1.8.patch uploaded by rmudgett (License #5621)
(closes issue ASTERISK-18019)
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r323670 | rmudgett | 2011-06-15 11:43:31 -0500 (Wed, 15 Jun 2011) | 7 lines
Add a test to the event unit tests to catch ASTERISK-18002.
The new tests check to see if there are ANY subscribers to the event type
when ast_event_check_subscriber() is not passed any specific ie values.
(issue ASTERISK-18002)
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r323608 | seanbright | 2011-06-15 11:31:53 -0400 (Wed, 15 Jun 2011) | 39 lines
Merged revisions 323579 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r323579 | seanbright | 2011-06-15 11:22:50 -0400 (Wed, 15 Jun 2011) | 32 lines
Merged revisions 323559 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r323559 | seanbright | 2011-06-15 11:15:30 -0400 (Wed, 15 Jun 2011) | 25 lines
Resolve a segfault/bus error when we try to map memory that falls on a page
boundary.
The fix for ASTERISK-15359 was incorrect in that it added 1 to the length of the
mmap'd region. The problem with this is that reading/writing to that extra byte
outside of the bounds of the underlying fd causes a bus error.
The real issue is that we are working with both a FILE * and the raw fd
underneath it and not synchronizing between them. The code that was removed in
ASTERISK-15359 was correct, but we weren't flushing the FILE * before mapping
the fd.
Looking at the manager code in 1.4 reveals that the FILE * in 'struct
mansession' is never used except to create a temporary file that we immediately
fdopen. This means we just need to write a 0 byte to the fd and everything will
just work. The other branches require a call to fflush() which, while not a
guaranteed fix, should reduce the likelihood of a crash.
This all makes sense in my head.
(closes issue ASTERISK-16460)
Reported by: Ravelomanantsoa Hoby (hoby)
Patches:
issue17747_1.4_svn_markII.patch uploaded by Sean Bright (license #5060)
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r323371 | jrose | 2011-06-14 11:38:43 -0500 (Tue, 14 Jun 2011) | 12 lines
Changes contact use in build_peer to use the FORCE_RPORT flag instead of RPORT_PRESENT
It turned out that this was causing NAT=Yes to always use rport when present which was
against 1.6.2 behavior and the check itself was redundant since the only way this
segment of code could be reached was if RPORT_PRESENT was already evaluated as true
earlier.
(closes issue ASTERISK-17789)
Reported by: byronclark
Patches:
use_sip_nat_force_rport.patch uploaded by byronclark (license 1200)
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Current behavior in ConfBridge menu definitions is that first definition takes
precedence, even in templated situations. This change allows inheritance and
overriding to work as expected so that the last definition takes precedence.
(closes ASTERISK-17986)
Review: https://reviewboard.asterisk.org/r/1267/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323272 f38db490-d61c-443f-a65b-d21fe96a405b
This adds the playing_moh flag to the conference_bridge_user struct that
signifies when MOH should be playing so code doesn't have to guess whether
MOH is playing.
This change also adds the necessary checking to ensure that MOH continues
playing for a single user in a conference after the join sound is played when
configured to do so.
(closes ASTERISK-17988)
Review: https://reviewboard.asterisk.org/r/1263/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323107 f38db490-d61c-443f-a65b-d21fe96a405b
Bridge and user profiles are not checked for existence before use. The lack
of a fully formed bridge profile can cause a segfault when sounds are accessed.
This change ensures that bridge and user profiles exist prior to usage
attempts.
Review: https://reviewboard.asterisk.org/r/1264/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323106 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r323040 | mnicholson | 2011-06-10 14:20:41 -0500 (Fri, 10 Jun 2011) | 5 lines
Unlock the sip channel during fax detection like chan_dahdi does to prevent a deadlock with ast_autoservice_stop.
(closes issue ASTERISK-17798)
tested by mnicholson
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r322981 | twilson | 2011-06-10 08:29:00 -0700 (Fri, 10 Jun 2011) | 11 lines
Avoid a DB1 infinite loop bug
Explicity check the last entry in the DB and make sure that we don't iterate
past it. Since there can be no duplicates, this just makes sure that we stop
after matching the last key.
This patch also refactors the code to get away from some code duplication. A
previous patch added many astdb tests and this patch passed them.
Review: https://reviewboard.asterisk.org/r/1259/
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r322749 | rmudgett | 2011-06-09 11:31:53 -0500 (Thu, 09 Jun 2011) | 15 lines
Remove potential deadlock in call pickup race.
Deadlock is possible in ast_do_pickup() when holding the target channel
lock and trying to get the chan channel lock. Also, holding the target
lock when calling ast_channel_masquerade() is not a good idea because that
routine does deadlock avoidance.
* Removed the need to hold the target lock after marking the target with a
datastore and getting the connected line data off of the target channel.
* Moved can_pickup() to ast_can_pickup() in features.c. Now all the call
pickup methods use the same basic call pickup availability check.
Review: https://reviewboard.asterisk.org/r/1234/
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r322585 | jrose | 2011-06-09 09:06:42 -0500 (Thu, 09 Jun 2011) | 11 lines
Adds ast_escape_encoded utility to properly handle escaping of quoted field before uri.
This commit backports a feature in trunk affecting initreqprep so that display name won't
be encoded improperly. Also includes unit tests for the ast_escape_quoted function.
This patch gives 1.8 a much improved outlook in countries which don't use standard
ASCII characters.
(closes issue ASTERISK-16949)
Reported by: Örn Arnarson
Review: https://reviewboard.asterisk.org/r/1235/
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Autoanswer added to skinny based on incoming chan var SKINNY_AUTOANSWER.
Initial value must be the time to autoanswer in ms, then optionally :BEEP
to play a tone when answered and :MUTE to mute the mic when answering.
eg 3000:MUTE:BEEP will ring for 3 secs, then answer, mute the mic, and
play a beep. just 3000 would answer afer 3 secs of ringing with no
beep and full two way audio.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@322544 f38db490-d61c-443f-a65b-d21fe96a405b
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r322484 | rmudgett | 2011-06-08 15:46:55 -0500 (Wed, 08 Jun 2011) | 15 lines
Ring all queue with more than 255 agents will cause crash.
1. Create a ring-all queue with 500 permanent agents.
2. Call it.
3. Asterisk will crash.
The watchers array in app_queue.c has a hard limit of 255. Bounds
checking is not done on this array. No sane person should put 255 people
in a ring-all queue, but we should not crash anyway.
* Added bounds checking to the watchers array.
JIRA AST-464
JIRA SWP-2903
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r322425 | rmudgett | 2011-06-08 13:46:30 -0500 (Wed, 08 Jun 2011) | 16 lines
SRV lookup attempted for SIP peers listed as an IP address.
Asterisk attempts to SRV lookup a host name even if the host name is an IP
address. Regression introduced when IPv6 support was added.
* Restored the check in ast_dnsmgr_lookup() to see if the given host name
is an IP address. The IP address could be in either IPv4 or IPv6 formats.
(closes issue ASTERISK-17815)
Reported by: Byron Clark
Tested by: Byron Clark, Richard Mudgett
Patches:
issue19248_v1.8.patch - uploaded by Richard Mudgett (License #5621)
Review: https://reviewboard.asterisk.org/r/1240/
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The do_monitor seemed to be there for task scheduling and network monitoring. However, the network monitoring has a dedicated thread so the ast_io_wait was basically just a usleep as it didn't actually seem to be monitoring anything.
Review: https://reviewboard.asterisk.org/r/1256/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@322381 f38db490-d61c-443f-a65b-d21fe96a405b
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r322322 | irroot | 2011-06-08 08:18:38 +0200 (Wed, 08 Jun 2011) | 18 lines
Make handle_request_publish do dialog expiration and destruction.
This patch fixes handle_request_publish so that it does dialog expiration and destruction.
Without this patch the incoming PUBLISH requests will get stuck in the dialog list.
Restarting asterisk is the only way to remove them.
Personal observation on one system the server hung up while looping through the channels
rendering asterisk unusable and all sip phones unregisterd when they try reregister
more requests are added.
(closes issue #18898)
Reported by: gareth
Tested by: loloski, Chainsaw, wimpy, se, kuj, irroot
Jira: https://issues.asterisk.org/jira/browse/ASTERISK-17915
Review: https://reviewboard.asterisk.org/r/1253
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r321926 | rmudgett | 2011-06-03 17:09:36 -0500 (Fri, 03 Jun 2011) | 18 lines
Asterisk crash when unloading cdr_radius/cel_radius.
The rc_openlog() API call is passed a string that is used by openlog() to
format log messages. The openlog() does not copy the string it just keeps
a pointer to it. When the module is unloaded, the string is gone from
memory. Depending upon module load order and if the other module then has
an error, a crash happens.
* Pass rc_openlog() a strdup'd string with the understanding that there
will be a small memory leak if the cdr_radius/cel_radius modules are
unloaded.
* Call rc_destroy() to free the rc handle memory when the module is
unloaded.
JIRA AST-483
JIRA SWP-3062
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r321924 | rmudgett | 2011-06-03 16:49:17 -0500 (Fri, 03 Jun 2011) | 5 lines
Be more explicit for CCSS generic device state event subscription.
Make CCSS generic device state event subscription specify the
AST_EVENT_IE_STATE ie exists to be safe.
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r321871 | rmudgett | 2011-06-03 15:58:13 -0500 (Fri, 03 Jun 2011) | 27 lines
Event subscription fixes.
Must commit the subscription fixes together with the integration
subscription tests. The subscription fixes cause an erroneously passing
test to fail. The new subscription tests detect errors without the
subscription fixes.
* Added missing event_names[] table entry.
* Reworked ast_event_check_subscriber()/match_sub_ie_val_to_event() to
correctly detect if a subscriber exists for the proposed event.
* Made match_ie_val() and match_sub_ie_val_to_event() check the buffer
length for RAW payload types.
* Fixed error handling memory leak in ast_event_sub_activate(),
ast_event_unsubscribe(), and ast_event_queue().
* Made ast_event_new() and ast_event_check_subscriber() better protect
themselves from an invalid payload type.
* Added container lock protection between removing old cache events and
adding the new cached event in
ast_event_queue_and_cache()/event_update_cache().
* Added new event subscription tests.
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r321753 | russell | 2011-06-03 13:32:45 -0500 (Fri, 03 Jun 2011) | 2 lines
Backport an astobj2 unit test so that it runs on 1.8 as well.
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Don't send all messages to 's'. Get the destination from the request URI.
(Found using automated test cases).
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@321617 f38db490-d61c-443f-a65b-d21fe96a405b
Asterisk now has protocol independent support for processing text messages
outside of a call. Messages are routed through the Asterisk dialplan.
SIP MESSAGE and XMPP are currently supported. There are options in sip.conf
and jabber.conf that enable these features.
There is a new application, MessageSend(). There are two new functions,
MESSAGE() and MESSAGE_DATA(). Documentation will be available on
the project wiki, wiki.asterisk.org.
Thanks to Terry Wilson for the assistance with development and to David Vossel
for helping with some additional testing.
Review: https://reviewboard.asterisk.org/r/1042/
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r321528 | may | 2011-06-01 14:40:19 +0400 (Wed, 01 Jun 2011) | 14 lines
Fix double alerting, add forced alerting before answer
Fix double alerting (it wasn't fixed here by issue #18542)
Add forced alerting before connect (if it wasn't before)
Try to send all packets from outgoing queue rather than one only
Call goes into clearing state when disconnect command is received
(closes issue #19361)
Reported by: vmikhelson
Patches:
issue19361-3.patch uploaded by may213 (license 454)
Tested by: vmikhelson
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r321392 | rmudgett | 2011-05-27 18:45:41 -0500 (Fri, 27 May 2011) | 12 lines
Crash when using hagi and no servers are available.
When none of the servers returned by the SRV querey respond, asterisk
crashes. The problem is that if the loop over all the SRV entries
finishes then the srv_context has already been cleaned up.
* Make ast_srv_cleanup() check to see if the context is already cleaned
up.
(closes issue #19256)
Reported by: byronclark
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r321330 | rmudgett | 2011-05-27 16:31:25 -0500 (Fri, 27 May 2011) | 8 lines
The app_privacy args have undocumented "options" position, interferes with "context" position.
* Add documention for unused "options" position to match existing code.
The trunk(v1.10) version will remove the unused options position.
(closes issue #19273)
Reported by: mdavenport
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r321211 | alecdavis | 2011-05-27 20:31:15 +1200 (Fri, 27 May 2011) | 16 lines
Fix *8 directed pickup locks system during pickupsound play out
move playout from sip_pickup_thread to bridge using BRIDGE_PLAY_SOUND method,
This stop the clash of 2 threads trying to write audio to same channel.
In addition fixes choppy audio beep in issue 19177.
(issue #18654)
(issue #19177)
Reported by: Docent
Patches:
review1232-1.8.diff.txt alecdavis (license 585)
Tested by: alecdavis
Review: https://reviewboard.asterisk.org/r/1232/
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r321155 | markm | 2011-05-26 17:48:45 -0400 (Thu, 26 May 2011) | 10 lines
Fixed build problem with dev mode enabled, which was caused by commit 321100. Reformulated patch to be more generic.
Moved the sip uri parse variable initalization to parse_uri_full in reqresp_parser.c. This will ensure that any use of parse uri will have null output variables if the parse fails.
(closes issue #19346)
Reported by: kobaz
Tested by: kobaz,JonathanRose
Review: [full review board URL with trailing slash]
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r321100 | markm | 2011-05-26 16:09:35 -0400 (Thu, 26 May 2011) | 11 lines
ast_sockaddr_resolve() in netsock2.c may deref a null pointer
Added a null check in netsock2 ast_sockaddr_resolve() as well as added default initalizers in chan_sip parse_uri_legacy_check() to make sure that invalid uris will make null (and not undefined) user,pass,domain,transport variables
(closes issue #19346)
Reported by: kobaz
Patches:
netsock2.patch uploaded by kobaz (license 834)
Tested by: kobaz, Marquis
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r321042 | twilson | 2011-05-26 10:29:54 -0700 (Thu, 26 May 2011) | 6 lines
Initialize stack-allocated ast_sockaddrs before use
It is important to always initialize ast_sockaddrs before use--even if they
are passed to ast_sockaddr_copy as the underlying storage could be bigger
than what ends up being copied--leaving part of the data unitialized.
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The ast_string_field_build_va functions were written to take to separate
va_lists to work around FreeBSD 4 not having va_copy defined.
In the end, we don't support anything using gcc < 3 anyway because we use
va_copy all over the place anyway. This patch just simplifies things by
removing the second va_list function arguments in favor of va_copy.
Review: https://reviewboard.asterisk.org/r/1233/
--This line, and those below, will be ignored--
M include/asterisk/stringfields.h
M main/utils.c
M main/channel.c
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r320883 | rmudgett | 2011-05-25 17:25:18 -0500 (Wed, 25 May 2011) | 17 lines
Native SIP CCSS sends bad CC cancel SUBSCRIBE message.
The SUBSCRIBE message used to cancel a CC request has incorrect To/From
SIP headers. They are reversed and the dialog tags are the same when they
should not be. If pedantic mode was disabled, then the cancel would have
succeeded despite the incorrect message.
* The SIP_OUTGOING flag was not set correctly for the dialog and I had to
move some CC subscribe handling code as a result.
* Initialized the dialog subscribed type to CALL_COMPLETION earlier. If a
CC request SUBSCRIBE message comes in and the CC instance is not found,
the 404 response was duplicated.
JIRA AST-568
JIRA SWP-3493
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r320823 | rmudgett | 2011-05-25 12:06:38 -0500 (Wed, 25 May 2011) | 18 lines
The AMI Newstate event contains different information between v1.4 and v1.8.
The addition of connected line support in v1.8 changes the behavior of the
channel caller ID somewhat. The channel caller ID value no longer time
shares with the connected line ID on outgoing call legs. The timing of
some AMI events/responses output the connected line ID as caller ID.
These party ID's are now separate.
* The ConnectedLineNum and ConnectedLineName headers were added to many
AMI events/responses if the CallerIDNum/CallerIDName headers were also
present.
(closes issue #18252)
Reported by: gje
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1227/
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r320796 | rmudgett | 2011-05-25 11:23:11 -0500 (Wed, 25 May 2011) | 17 lines
Give zombies a safe channel driver to use.
Recent crashes from zombie channels suggests that they need a safe home to
goto. When a masquerade happens, the physical part of the zombie channel
is hungup. The hangup normally sets the channel private pointer to NULL.
If someone then blindly does a callback to the channel driver, a crash is
likely because the private pointer is NULL.
The masquerade now sets the channel technology of zombie channels to the
kill channel driver.
Related to the following issues:
(issue #19116)
(issue #19310)
Review: https://reviewboard.asterisk.org/r/1224/
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r320650 | rmudgett | 2011-05-23 12:53:44 -0500 (Mon, 23 May 2011) | 16 lines
Add ConnectedLineNum/Name headers to output of AMI action Status.
* Add ConnectedLineNum and ConnectedLineName headers to the output of the
AMI action Status. This makes it easier to find out who the channel is
connected to without having to lookup BridgedChannel or when they are
connected to an application (e.g.: VoiceMail) which has no bridged
channel.
* Bridged channels with no CallerID had "" instead of "<unknown>" output,
that might be a bug as "<unknown>" was what older versions used.
(closes issue #18158)
Reported by: gareth
Patches:
svn-292308.diff uploaded by gareth (license 208)
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r320180 | mnicholson | 2011-05-20 13:48:46 -0500 (Fri, 20 May 2011) | 16 lines
This commit modifies the way polling is done on TLS sockets.
Because of the buffering the TLS layer does, polling is unreliable. If poll is
called while there is data waiting to be read in the TLS layer but not at the
network layer, the messaging processing engine will not proceed until something
else writes data to the socket, which may not occur. This change modifies the
logic around TLS sockets to only poll after a failed read on a non-blocking
socket. This way we know that there is no data waiting to be read from the
buffering layer.
(closes issue #19182)
Reported by: st
Patches:
ssl-poll-fix3.diff uploaded by mnicholson (license 96)
Tested by: mnicholson
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r320162 | jrose | 2011-05-20 13:12:21 -0500 (Fri, 20 May 2011) | 15 lines
Fixes an imapfolder related crash
imapfolders being set in the general section of voicemail would cause the inbox folder name to
change. Since sound file names are made based on the names of the folders, this would cause
the audio related to that folder name to change and if Asterisk attempted to play it, the
channel would instantly hang up when the audio file couldn't be found. This patch searches for
the name of the folder first to leave existing behavior in tact and if that fails, it uses
the normal inbox name to get the sound file instead.
(closes issue #16104)
Reported by: blkline
Review: https://reviewboard.asterisk.org/r/1215/
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r320057 | rmudgett | 2011-05-20 11:43:02 -0500 (Fri, 20 May 2011) | 19 lines
Crash while transferring a call during DTMF feature timeout.
When a call is being attended transferred during the time between
AST_FRAME_DTMF_BEGIN and AST_FRAME_DTMF_END, the transferred channel
becomes a zombie (so tech data is not available), making ast_dtmf_stream()
segfault when it tries to send the DTMF digit (at least with SIP
channels).
Patch based on feature-end-zombie.patch uploaded by Irontec (license 1256)
* Check for zombies when ast_channel_bridge() returns.
* Guarantee that the fo parameter value is initialized in
ast_channel_bridge() before any returns.
(closes issue #19116)
Reported by: Irontec
Tested by: rmudgett
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r319997 | rmudgett | 2011-05-20 10:48:25 -0500 (Fri, 20 May 2011) | 25 lines
Crash when using directed pickup applications.
The directed pickup applications can cause a crash if the pickup was
successful because the dialplan keeps executing.
This patch does the following:
* Completes the channel masquerade on a successful pickup before the
application returns. The channel is now guaranteed a zombie and must not
continue executing the dialplan.
* Changes the return value of the directed pickup applications to return
zero if the pickup failed and nonzero(-1) if the pickup succeeded.
* Made some code optimizations that no longer require re-checking the
pickup channel to see if it is still available to pickup.
(closes issue #19310)
Reported by: remiq
Patches:
issue19310_v1.8_v2.patch uploaded by rmudgett (license 664)
Tested by: alecdavis, remiq, rmudgett
Review: https://reviewboard.asterisk.org/r/1221/
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r319938 | jrose | 2011-05-20 08:28:24 -0500 (Fri, 20 May 2011) | 12 lines
Adds legacy_useroption_parsing to address interoperability concerns.
With the new option engaged, Asterisk should interpret user fields with useroptions
contained within the userfield of the uri by stripping them out of the original message
whenever a semicolon is encountered in the userfield string.
(closes issue #18344)
Reported by: danimal
Tested by: jrose
Review: https://reviewboard.asterisk.org/r/1223/
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r319758 | rmudgett | 2011-05-19 11:50:48 -0500 (Thu, 19 May 2011) | 21 lines
CCSS generic agent with POTS and ISDN phones fail caller busy call-back test.
If the following is true after a CCSS activation:
* The generic agent is for an analog phone or ISDN phone. (Caller party)
* The called party becomes available.
* The caller party is not available.
When the caller party becomes available, the caller is not alerted to the
called party being available. The generic agent still thinks the caller
is busy.
* Fixed the generic agent device state event subscription to look for all
device states that are considered available.
* Encapsulated the device state test for CCSS generic device available in
cc_generic_is_device_available(). Made the generic agent and monitor use
the new function instead of the manually coded inline equivalent.
JIRA AST-559
JIRA SWP-3462
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r319654 | twilson | 2011-05-18 16:15:58 -0700 (Wed, 18 May 2011) | 22 lines
Merged revisions 319653 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r319653 | twilson | 2011-05-18 16:11:57 -0700 (Wed, 18 May 2011) | 15 lines
Merged revisions 319652 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r319652 | twilson | 2011-05-18 16:04:35 -0700 (Wed, 18 May 2011) | 8 lines
Make sure everyone gets an unhold when a transfer succeeds
Some phones, like the Snom phones, send a hold to the transfer target after
before sending the REFER. We need to make sure that we unhold the parties
that are being connected after the masquerade. If Local channels with the /nm
option are used when dialing the parties, hold music would still be playing on
the transfer target, even after being connected with the transferee.
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r319552 | twilson | 2011-05-18 13:22:36 -0700 (Wed, 18 May 2011) | 11 lines
Unbreak the storing of registrations for restart
The fix for issue 18882 broke retrieving non-realtime peers from the ast_db
on restart/reload. This patch tries to unbreak things while leaving the intent
of the original fix intact.
(closes issue #19318)
Reported by: remiq
Patches:
diff.txt uploaded by twilson (license 396)
Tested by: lmadsen, remiq
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r319529 | twilson | 2011-05-18 13:05:34 -0700 (Wed, 18 May 2011) | 24 lines
Merged revisions 319528 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r319528 | twilson | 2011-05-18 13:02:06 -0700 (Wed, 18 May 2011) | 17 lines
Merged revisions 319527 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r319527 | twilson | 2011-05-18 12:56:08 -0700 (Wed, 18 May 2011) | 10 lines
Fix app_dial ring groups
Revert part of r315643. We need to remove the datastore here as well.
The code in bridging code will catch anything that app_dial might miss.
(closes issue #19311)
Reported by: mspuhler
Patches:
issue_19311_no_answer.diff uploaded by elguero (license 37)
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r319469 | rmudgett | 2011-05-17 16:57:56 -0500 (Tue, 17 May 2011) | 22 lines
Merged revision 319468 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
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r319468 | rmudgett | 2011-05-17 16:49:31 -0500 (Tue, 17 May 2011) | 15 lines
The mISDN HDLC mode is prevented on dialed channels.
The use of mISDN HDLC mode is prevented if the mISDN dial technology
option 'h1' is used when config option astdtmf=yes.
There is a bug in channels/misdn/isdn_lib.c which prevents the use of HDLC
mode. Instead of setting the channel to HDLC mode it is set to
transparent(no dsp, no hdlc), although hdlc is not "no hdlc". I.e the
logging message is correct, but the if condition is not.
Make check the nodsp and hdlc flags.
JIRA ABE-2787
JIRA SWP-3437
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The NEC SV8300 rejects the Q931_IE_TIME_DATE for Q.SIG.
Add option to specify if and how much of the current time is put in
Q931_IE_TIME_DATE.
* Send date/time ie never.
* Send date/time ie date only.
* Send date/time ie date and hour.
* Send date/time ie date, hour, and minute.
* Send date/time ie date, hour, minute, and second.
* Send date/time ie default: Libpri will send date and hhmm only when in
NT PTMP mode to support ISDN phones.
(closes issue #19221)
Reported by: kenner
JIRA SWP-3396
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@319427 f38db490-d61c-443f-a65b-d21fe96a405b
Probably haven't been working for a couple of years. May still need
some more love, but they are now working, both as a hint device and
monitoring a hint. Changes centre around the long ago change
to remove the requirement for a device name in a skinny line, and
changes to the transmit_* functions.
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r319261 | jrose | 2011-05-16 16:00:55 -0500 (Mon, 16 May 2011) | 2 lines
Makes busy detection in dsp.c always allow for at least one frame (20ms) of error so that 200ms tone lengths don't get ignored by single frame error lengths.
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r319259 | rmudgett | 2011-05-16 15:33:37 -0500 (Mon, 16 May 2011) | 13 lines
Deadlock between generic CCSS agent and native ISDN CCSS.
Deadlock can occur when the generic CCSS agent is deleting duplicate CC
offers and the native ISDN CC driver is processing an incoming CC message.
The cc_core_instances container lock cannot be held when an agent or
monitor callback is invoked without the possibility of a deadlock.
* Make kill_duplicate_offers() remove the reference in cc_core_instances
outside of the container lock.
JIRA AST-566
JIRA SWP-3469
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state of the channel reverts to unknown this should be rejected.
this is important for negotiating T.38 gateway see #13405
This patch adds a option T38_REJECTED that behaves as T38_DISABLED except it reports state rejected.
Trivial Change to res_fax to honnor UNAVAILABLE and REJECTED states.
(closes issue #18889)
Reported by: irroot
Tested by: irroot, darkbasic, mnicholson
Review: https://reviewboard.asterisk.org/r/1115
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When called, activatesub first cleans up the active sub and then
handles the sub passed. dialandactivatesub first sets sub->exten
and then calls activatesub. Revise handle_offhook to utilise the
callid sent to chan_skinny. Some other minor fixes especially around
d->hookstate (which still needs some more work).
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r318919 | bbryant | 2011-05-13 14:04:50 -0400 (Fri, 13 May 2011) | 10 lines
This patch fixes an issue with SRTP which makes HOLD/UNHOLD impossible when too
much time has passed between sending audio.
(closes issue #18206)
Reported by: bernhardsi
Patches:
res_srtp_unhold.patch uploaded by bernhards (license 1138)
Tested by: bernhards, notthematrix
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r318868 | rmudgett | 2011-05-13 11:28:26 -0500 (Fri, 13 May 2011) | 19 lines
CDR's are being written immediately on caller hangup.
CDR's are being written immediately on caller hangup. The dialplan is not
able to modify it in the h exten. The h exten in the initial context is
not run before closing CDR's when the bridge is unlinked if a macro is
active and does not have an h exten.
* Make ast_bridge_call() check for an h exten in the current context and
if a macro is active then the initial context. The first h exten found is
then run before closing the CDR.
(closes issue #18212)
Reported by: leearcher
Patches:
issue18212_v1.8.patch uploaded by rmudgett (license 664)
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1206/
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There were some issues where if a simple switch was cancelled and a
new switch started before the first had timed out where the d->exten
would be used for both subchannels. This was bad leading to possible
invalid extensions if some digits had been entered in the abandoned
simple switch and the second one was completed before the first timed
out, or the second would be cancelled because d->exten would be set to
nothing on the time out of the first.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@318833 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r318783 | rmudgett | 2011-05-12 20:47:05 -0500 (Thu, 12 May 2011) | 14 lines
PRI early media won't ring.
And another way to pass early media. Don't indicate that there is inband
information present, just assume that the B channel is connected.
* Restore clearing the dialing flag Rx squelch unconditionally when a
PROCEEDING message comes in.
(closes issue #19268)
Reported by: tbsky
Patches:
issue19268_v1.8.patch uploaded by rmudgett (license 664)
Tested by: tbsky
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@318784 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r318671 | alecdavis | 2011-05-13 10:52:08 +1200 (Fri, 13 May 2011) | 30 lines
Fix directed group pickup feature code *8 with pickupsounds enabled
Since 1.6.2, the new pickupsound and pickupfailsound in features.conf cause many issues.
1). chan_sip:handle_request_invite() shouldn't be playing out the fail/success audio, as it has 'netlock' locked.
2). dialplan applications for directed_pickups shouldn't beep.
3). feature code for directed pickup should beep on success/failure if configured.
Created a sip_pickup() thread to handle the pickup and playout the audio, spawned from handle_request_invite.
Moved app_directed:pickup_do() to features:ast_do_pickup().
Functions below, all now use the new ast_do_pickup()
app_directed_pickup.c:
pickup_by_channel()
pickup_by_exten()
pickup_by_mark()
pickup_by_part()
features.c:
ast_pickup_call()
(closes issue #18654)
Reported by: Docent
Patches:
ast_do_pickup_1.8_trunk.diff.txt uploaded by alecdavis (license 585)
Tested by: lmadsen, francesco_r, amilcar, isis242, alecdavis, irroot, rymkus, loloski, rmudgett
Review: https://reviewboard.asterisk.org/r/1185/
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@318672 f38db490-d61c-443f-a65b-d21fe96a405b
Consolidate the functions and add some debugging info. Allows to be
able to set a substate without explicitly knowing what the state is.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@318635 f38db490-d61c-443f-a65b-d21fe96a405b
Add the setsubstate_onhook to complete the initial substate handling
procedures. Added dumpsub(sub, forcehangup) which is the common way of
calling setsubstate_onhook. Dumpsub attempts to activate another sub
after setting the current one onhook.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@318600 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r318549 | twilson | 2011-05-11 13:39:48 -0500 (Wed, 11 May 2011) | 27 lines
Merged revisions 318548 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r318548 | twilson | 2011-05-11 12:15:39 -0500 (Wed, 11 May 2011) | 19 lines
Clean up several chan_sip reference leaks
Several situations in the code could lead to peers or sip_pvt references
being leaked. This would cause RTP ports to never be destroyed (leading
to exhaustion of all available RTP ports) and memory leaks.
The original patch for this issue from rgagnon was the result of an
obscene amount of testing and hard work, for which I am very grateful. I
did some cleanup and added a few additional refcount fixes that I found.
(closes issue #17255)
Reported by: kvveltho
Patches:
tag-1.6.2.17-r309252-sip-dos-mem-leak-fix.diff uploaded by rgagnon (license 1202)
Tested by: rgagnon, twilson, wdoekes, loloski
Review: https://reviewboard.asterisk.org/r/1101/
Review: https://reviewboard.asterisk.org/r/1207/
Review: https://reviewboard.asterisk.org/r/1210/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@318551 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r318499 | rmudgett | 2011-05-10 18:41:08 -0500 (Tue, 10 May 2011) | 15 lines
Unable to pickup DAHDI/PRI call because call state is reported as DIALING.
The channel state is not updated to RINGING when an ALERTING message is
received. Regression caused when sig_pri.c (also sig_ss7.c) extracted
from chan_dahdi.c.
* Added missing channel state update to RINGING when the
AST_CONTROL_RINGING frame is queued for ISDN and SS7.
(closes issue #19257)
Reported by: alecdavis
Patches:
issue19257_v1.8_v2.patch uploaded by rmudgett (license 664)
Tested by: alecdavis, rmudgett
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@318500 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r318337 | twilson | 2011-05-09 15:23:15 -0500 (Mon, 09 May 2011) | 18 lines
Merged revisions 318331 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r318331 | twilson | 2011-05-09 15:04:41 -0500 (Mon, 09 May 2011) | 12 lines
Don't offer video to directmedia callee unless caller offered it as well
Make sure that when directmedia is enabled, that video is not offered to the
callee even if it supports it. p->vrtp will not exist since the caller didn't
offer video.
(closes issue #19195)
Reported by: one47
Patches:
sip_cant_add_video_rtp uploaded by one47 (license 23)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@318400 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r318351 | rmudgett | 2011-05-09 18:15:32 -0500 (Mon, 09 May 2011) | 6 lines
Remove references to res_features and its export file.
The contents of res/res_features.c was moved to into main/features.c
awhile ago. There is no longer any need for the res/Makefile to reference
res_features or the res_features linker exports file to exist.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@318352 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r318282 | rmudgett | 2011-05-09 14:07:01 -0500 (Mon, 09 May 2011) | 24 lines
Hangup extension executed twice.
When a user hangs up a call, in certain circumstances, the hangup
extension can end up being executed twice:
1) If a call is bridged and the 'h' extension executes the Hangup
application, then the 'h' extension will be executed twice.
2) If a call is bridged within a macro (Dial or Queue), it has its own 'h'
extension, the main context also has an 'h' extension, and the macro 'h'
extension executes the Hangup application, then both 'h' extensions will
be executed.
* Revert originally commited fix for #16106 and just set
AST_FLAG_BRIDGE_HANGUP_RUN unconditionally in ast_bridge_call(). The
bridge code just executed an 'h' extension so the main PBX loop does not
need to execute one as well.
(issue #16106)
Reported by: ajohnson
(issue #16548)
Reported by: hajekd
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@318283 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r318231 | rmudgett | 2011-05-09 11:57:18 -0500 (Mon, 09 May 2011) | 41 lines
Don't get early media for ISDN on outgoing calls.
It looks to be a long-standing misinterpretation of the progress indicator
ie values:
1 - Call is not end-to-end ISDN; further call progress information may be
available in-band.
8 - In-band information or an appropriate pattern is now available.
Only value 8 is handled by chan_dahdi/sig_pri. The 1 value is not handled
as early media probably because the meaning of the second half of it's
description was overlooked.
* Test to see if either PRI_PROG_CALL_NOT_E2E_ISDN(1) or
PRI_PROG_INBAND_AVAILABLE(8) bits are set to open the media path.
(closes issue #18868)
Reported by: isrl
Patches:
issue18868_19246_v1.8.patch uploaded by rmudgett (license 664)
Tested by: satish_lx
..........
No inband progress on PRI_EVENT_RINGING even if inband flag set.
My ISDN-PRI provider sends an ALERTING with "Inband information or
appropriate pattern now available", but Asterisk only generates and passes
the RING to the SIP extension, not the inband message. Unfortunately, the
inband message is not a ringback tone but a prompt that says the number is
not in service. The SIP extension then hears two rings and the call is
hungup which confuses the caller.
* Post an AST_CONTROL_PROGRESS as well as opening the media path if inband
audio is indicated with an ALERTING message.
(closes issue #19246)
Reported by: cristiandimache
Patches:
issue19246_v1.8.patch uploaded by rmudgett (license 664)
Tested by: cristiandimache
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@318232 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r318148 | jrose | 2011-05-09 09:18:14 -0500 (Mon, 09 May 2011) | 4 lines
Documenting an observed behavior of features in features.conf. Since parkinglots use an
integer for the parkinglot extensions, leading zeros specified in the configuration file
are ignored.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@318162 f38db490-d61c-443f-a65b-d21fe96a405b
If a call is made to a line that already has a call and the device is
offhook (ie activeish call), the call is set to CALLWAIT rather than RINGIN.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@318106 f38db490-d61c-443f-a65b-d21fe96a405b
Long time coming, finally moving the hookstate from line to device.
This may fix some issues where a device has multiple lines. Previously
we had to run through all lines on a device to see if it was actually
onhook or not.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317996 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r317969 | russell | 2011-05-06 16:49:01 -0500 (Fri, 06 May 2011) | 10 lines
Use the right variable to print the time in a debug message.
The original patch also increased some buffer sizes, but that was already
done in this version.
(closes issue #17034)
Reported by: sysreq
Patches:
asterisk-issue-17034.patch uploaded by sysreq (license 1009)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317970 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r317867 | russell | 2011-05-06 15:01:16 -0500 (Fri, 06 May 2011) | 10 lines
chan_sip: Destroy variables on a sip_pvt before copying vars from the sip_peer.
Don't duplicate variables on the sip_pvt. Just reset the variable list each
time.
(closes issue #19202)
Reported by: wdoekes
Patches:
issue19202_destroy_challenged_invite_chanvars.patch uploaded by wdoekes (license 717)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317868 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r317865 | russell | 2011-05-06 14:46:49 -0500 (Fri, 06 May 2011) | 11 lines
chan_sip: fix a deadlock in check_rtp_timeout.
Don't block doing silly deadlock avoidance. Just return and try again later.
The funciton gets called often enough that it's fine. Also, this change was
already made in trunk.
(closes issue #18791)
Reported by: irroot
Patches:
chan_sip.rtptimeout.patch uploaded by irroot (license 52)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317866 f38db490-d61c-443f-a65b-d21fe96a405b
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r317861 | russell | 2011-05-06 14:35:00 -0500 (Fri, 06 May 2011) | 11 lines
URI encode less characters in the RPID and Contact headers.
If this change causes any problems, we will need to backport the more extensive
uri encoding and decoding handling changes that are in trunk/1.10.
(closes issue #18686)
Reported by: wolfgang
Patches:
quick-and-dirty.patch uploaded by wdoekes (license 717)
Tested by: wdoekes, devellow, wolfgang, mav3rick
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317862 f38db490-d61c-443f-a65b-d21fe96a405b
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r317858 | mnicholson | 2011-05-06 14:31:50 -0500 (Fri, 06 May 2011) | 6 lines
pbx_lua autoservice fixes
Don't start an autoservice in pbx_lua if pbx_lua already started one and don't
stop one if we didn't start one. Also start and stop the autoservice when
transferring control from and to the pbx.
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This change is already implemented in trunk.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317860 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r317837 | russell | 2011-05-06 14:24:11 -0500 (Fri, 06 May 2011) | 11 lines
Fix a crash in the MySQL() application.
This code was not handling channel datastores safely. The channel
must be locked.
(closes issue #17964)
Reported by: wuwu
Patches:
issue17964_addon_1.6.2_svn.patch uploaded by seanbright (license 71)
Tested by: wuwu
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317843 f38db490-d61c-443f-a65b-d21fe96a405b
automatically stopped when applications are executed, so this shouldn't cause
any problems.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317806 f38db490-d61c-443f-a65b-d21fe96a405b
Make autoservice_start() and autoservice_stop() return nothing. Also check if
the autoservice flag is set before starting or stopping the autoservice and
stop and start the autoservice when returning control to and getting control
from the pbx engine.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317803 f38db490-d61c-443f-a65b-d21fe96a405b
This code will actually detect any dialplan jump from any application that
calls ast_explicit_goto(). This change is only being done in trunk as it may
change the way some dialplans execute.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317721 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r317670 | rmudgett | 2011-05-06 11:19:18 -0500 (Fri, 06 May 2011) | 22 lines
Fix SIP connected line updates.
This patch fixes a couple SIP connected line update problems:
1) The connected line needs to be updated when the initial INVITE is sent
if there is a peer callerid configured. Previously, the connected line
information did not get reported until the call was connected so SIP could
not report connected line information in ringing or progress messages.
2) The connected line should not be updated on initial connect if there is
no connected line information. Previously, all it did was wipe out any
default preset CONNECTEDLINE information set by the dialplan with empty
strings.
(closes issue #18367)
Reported by: GeorgeKonopacki
Patches:
issue18367_v1.8.patch uploaded by rmudgett (license 664)
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1199/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317671 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r317584 | twilson | 2011-05-06 01:18:53 -0700 (Fri, 06 May 2011) | 20 lines
Merged revisions 317575 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r317575 | twilson | 2011-05-06 01:04:17 -0700 (Fri, 06 May 2011) | 13 lines
Merged revisions 317574 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r317574 | twilson | 2011-05-06 00:55:21 -0700 (Fri, 06 May 2011) | 6 lines
Re-fix queue round-robin
This part of the change for r315596 was incorrect. No bridge occurs
when doing a roundrobin dial and no one answers, so this code shouldn't
have been removed.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317596 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r317530 | russell | 2011-05-05 18:46:54 -0500 (Thu, 05 May 2011) | 10 lines
If the configure script runs, force a rebuild of menuselect-tree.
Some contents in the menuselect tree are dependent on configure script
parameters, namely --enable-dev-mode.
(closes issue #17219)
Reported by: Nick_Lewis
Patches:
issue_17219.rev1.txt uploaded by russell (license 2)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317531 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r317478 | russell | 2011-05-05 17:53:45 -0500 (Thu, 05 May 2011) | 12 lines
Fix some consistency issues with jitterbuffer config.
Store the defaults noted in the sample config files in the jitterbuffer config
data structure. This makes the CLI commands that output these settings show
the right thing. Also only show the settings that are relevant in the settings
CLI commands, based on which jitterbuffer is selected and whether it's enabled.
(closes issue #19083)
Reported by: rgagnon
Patches:
issue-19083-trunk-r313139.diff uploaded by rgagnon (license 1202)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317479 f38db490-d61c-443f-a65b-d21fe96a405b
I excluded the part of this patch that used the HOME environment variable since
the built-in editline library goes to great lengths to disallow that. Instead
only settings the EDITRC environment variable will use a user specified file.
Also, the default environment variable use to determine the edit more is
AST_EDITMODE instead of AST_EDITOR (although the latter is still supported).
(closes issue #15929)
Reported by: kkm
Patches:
astcli-editrc-v2.diff uploaded by kkm (license 888)
015929-astcli-editrc-trunk.240324.diff uploaded by kkm (license 888)
Tested by: seanbright
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317395 f38db490-d61c-443f-a65b-d21fe96a405b
skinny_hold moved to setsubstate_hold and skinny_unhold integrated into
setsubstate_connected. Removed sub->onhold and replaced with
SUBSTATE_HOLD.
Also fixed inbound call answering by queueing an AST_CONTROL_ANSWER on
answering a SUBSTATE_RINGIN sub (was a typo).
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317382 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r317370 | seanbright | 2011-05-05 16:25:52 -0400 (Thu, 05 May 2011) | 10 lines
Don't duplicate our data on the stack and just use the MYSQL_ROW directly.
With large result sets we were blowing out the stack.
(closes issue #19090)
Reported by: mickecarlsson
Patches:
issue19090_trunk_svn.patch uploaded by seanbright (license 71)
Tested by: mickecarlsson
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317377 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r317283 | jrose | 2011-05-05 14:09:13 -0500 (Thu, 05 May 2011) | 10 lines
Resolves a deadlock that occurs during sip_new
This is based on an uncommitted patch by jpeeler for the issue. Instead of
relocking and then unlocking the channel though, we keep the lock on the channel
until we are finished doing what we need to the channel.
(closes issue #18441)
Reported by: Alric
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317334 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r317281 | russell | 2011-05-05 13:39:44 -0500 (Thu, 05 May 2011) | 29 lines
Merged revisions 317255 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r317255 | russell | 2011-05-05 13:29:53 -0500 (Thu, 05 May 2011) | 22 lines
Merged revisions 317211 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r317211 | russell | 2011-05-05 13:20:29 -0500 (Thu, 05 May 2011) | 15 lines
chan_sip: fix broken realtime peer count, fix memory leak
This patch addresses two bugs in chan_sip:
1) The count of realtime peers and users was off. The increment checked the
value of the caching option, while the decrement did not.
2) Add a missing regfree() for a regex.
(closes issue #19108)
Reported by: vrban
Patches:
missing_regfree.patch uploaded by vrban (license 756)
sip_object_counter.patch uploaded by vrban (license 756)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317282 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r317196 | mnicholson | 2011-05-05 13:02:52 -0500 (Thu, 05 May 2011) | 8 lines
Set SO_KEEPALIVE on SIP TCP sockets so that they eventually go away when a peer
abruptly disappears. This mostly occurs after a successful registration.
(closes issue #17544)
Reported by: marcelloceschia
Patches:
(modified) tcptls.patch uploaded by st (license 907)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317198 f38db490-d61c-443f-a65b-d21fe96a405b
Move handling of both state handling from skinny_indicate to it's own sub.
Also, modified behaviour to not hangup the sub and let the dialplan
have a chance in doing what it wants for congestion. Added various states to
substate2str and added these states where applicable for other set_substate_
procs.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317026 f38db490-d61c-443f-a65b-d21fe96a405b
Move handling of setting busy state from skinny_indicate to it's own sub.
Also, modified behaviour to not hangup the sub and let the dialplan
have a chance in doing what it wants (eg busy(10); hangup() in the dialplan
now gives a busy indication for 10 secs and then hangs up.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@316994 f38db490-d61c-443f-a65b-d21fe96a405b
Moving a found object to the front of its bucket to reduce the necessary traversal steps to find an object. This change improves the search time on large system with many data or in link lists.
(closes issue #19233)
Reported by: schmidts
Review: https://reviewboard.asterisk.org/r/1201/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@316962 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r316917 | seanbright | 2011-05-04 22:23:28 -0400 (Wed, 04 May 2011) | 5 lines
Make sure that tcptls_session is properly initialized.
(issue #18598)
Reported by: ksn
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r316918 | seanbright | 2011-05-04 22:25:20 -0400 (Wed, 04 May 2011) | 5 lines
Look at the correct buffer for our digest info instead of an empty one.
(issue #18598)
Reported by: ksn
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r316919 | seanbright | 2011-05-04 22:30:45 -0400 (Wed, 04 May 2011) | 10 lines
Use the correct HTTP method when generating our digest, otherwise we always fail.
When calculating the 'A2' portion of our digest for verification, we need the
HTTP method that is currently in use. Unfortunately our mapping function was
incorrect, resulting in invalid hashes being generated and, in turn, failures
in authentication.
(closes issue #18598)
Reported by: ksn
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@316920 f38db490-d61c-443f-a65b-d21fe96a405b
Renamed previous setsubstate_ringout to setsubstate_dialing for a state
when attempting to dial a number, substate ringout now for when core
has indicated that the channel is actually ringing on the other end.
Also added substate2str for debugging purposes.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@316885 f38db490-d61c-443f-a65b-d21fe96a405b
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r316831 | rmudgett | 2011-05-04 13:51:40 -0500 (Wed, 04 May 2011) | 9 lines
Wait for leader with Music On Hold allows crosstalk between participants.
Parenthesis in the wrong position. Regression from issue #14365 when
expanding conference flags to use 64 bits.
(closes issue #18418)
Reported by: MrHanMan
Tested by: rmudgett
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@316832 f38db490-d61c-443f-a65b-d21fe96a405b
The functionality this patch attempts to achieve should already
be possible using [general](+) in the config file.
issue #17957
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r316617 | dvossel | 2011-05-04 08:44:41 -0500 (Wed, 04 May 2011) | 19 lines
Merged revisions 316616 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r316616 | dvossel | 2011-05-04 08:40:41 -0500 (Wed, 04 May 2011) | 12 lines
Fixes session-timers=refuse not being enforced for *caller*
During handle_request_invite, the session timer mode was retrieved from
a cached variable. This patch forces a peer lookup of the session timer
mode in the case of an incoming invite.
(closes issue #18804)
Reported by: wdoekes
Patches:
issue18804_session_timer_refuse_caller.patch uploaded by wdoekes (license 717)
issue_18804_v2.diff uploaded by dvossel (license 671)
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Added setsubstate_ringin. skinny_call now calls sss_ringin rather than inline.
Fixed previous issue so that setsubstate_connected now use SUBSTATE_RINGIN
to determine is an AST_CONTROL_ANSWER should be queued.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@316584 f38db490-d61c-443f-a65b-d21fe96a405b
Cosolidated the code so that skinny_answer now uses the setsubstate procedures
rather than doing the handling inline.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@316552 f38db490-d61c-443f-a65b-d21fe96a405b
Cosolidated the working out of the callinfo to be sent into
transmit_callinfo. Replaced ambiguous sub->outgoing with calldirection
which can be SKINNY_INCOMING or SKINNY_OUTGOING (same value as the
skinny protocol).
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@316519 f38db490-d61c-443f-a65b-d21fe96a405b
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r316476 | seanbright | 2011-05-03 22:34:01 -0400 (Tue, 03 May 2011) | 17 lines
Merged revisions 316475 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r316475 | seanbright | 2011-05-03 22:23:01 -0400 (Tue, 03 May 2011) | 10 lines
Honor the C option to MeetMe when L is passed.
This fixes a case that r304773 and friends missed.
(closes issue #17317)
Reported by: var
Patches:
meetme-continue-on-l_16218.diff uploaded by var (license 1227)
Tested by: seanbright
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The original implementation of this did the merging of all contexts with the
same name in the realtime layer, but that implementation severely breaks
drivers which use the same context name (e.g. iax.conf, type={peer,user}).
Therefore, the implementation needs to do the merging for particular entries
only, based upon what contexts would allow that in the channel driver itself.
This implementation is for chan_sip only, but others could be added in the
future.
(closes issue #17957)
Reported by: marcelloceschia
Patches:
chan-sip_parsing-general_branch162.patch uploaded by marcelloceschia (license 1079)
Tested by: tilghman
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@316428 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r316265 | russell | 2011-05-03 14:55:49 -0500 (Tue, 03 May 2011) | 5 lines
Fix a bunch of compiler warnings generated by gcc 4.6.0.
Most of these are -Wunused-but-set-variable, but there were a few others
mixed in here, as well.
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r316224 | rmudgett | 2011-05-03 14:18:30 -0500 (Tue, 03 May 2011) | 16 lines
The dahdi_hangup() call does not clean up the channel fully.
After dahdi_hangup() has supposedly hungup an ISDN channel there is still
traffic on the S0-bus because the channel was not cleaned up fully.
Shuffled the hangup code to include some missing cleanup. Also fixed some
code formatting in the area. I think the primary missing clean up code
was the call to tone_zone_play_tone() to turn off any active tones on the
channel.
(closes issue #19188)
Reported by: jg1234
Patches:
issue19188_v1.8.patch uploaded by rmudgett (license 664)
Tested by: jg1234
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r316217 | dvossel | 2011-05-03 13:59:06 -0500 (Tue, 03 May 2011) | 9 lines
Never put the Require: timer header in an Invite.
This has already been discussed and should have been resolved earlier. View
revsion 285565's log for more information about why it is important to not
put timer in the Require header.
(closes issue #18704)
Reported by: mfrager
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r315894 | mnicholson | 2011-04-27 14:14:27 -0500 (Wed, 27 Apr 2011) | 28 lines
Merged revisions 315893 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r315893 | mnicholson | 2011-04-27 14:03:05 -0500 (Wed, 27 Apr 2011) | 21 lines
Merged revisions 315891 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r315891 | mnicholson | 2011-04-27 13:57:56 -0500 (Wed, 27 Apr 2011) | 14 lines
Fix our compliance with RFC 3261 section 18.2.2.
This change optimizes the free_via() function and removes some redundant null
checking. It also fixes compliance with RFC 3261 section 18.2.2 by always using
the port specified in the Via header for routing responses (even when maddr is
not set). Also the htons() function is now used when setting the port.
Additional documentation comments have been added in various places to make the
logic in the code clearer.
(closes issue #18951)
Reported by: jmls
Patches:
issue18951_set_proper_port_from_via.patch uploaded by wdoekes (license 717) (modified)
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r315644 | twilson | 2011-04-26 14:39:01 -0700 (Tue, 26 Apr 2011) | 32 lines
Merged revisions 315643 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r315643 | twilson | 2011-04-26 14:27:44 -0700 (Tue, 26 Apr 2011) | 25 lines
Merged revisions 315596 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r315596 | twilson | 2011-04-26 14:16:10 -0700 (Tue, 26 Apr 2011) | 18 lines
Allow transfer loops without allowing forwarding loops
We try to avoid the situation where two phones may be forwarded to each other
causing an infinite loop by storing each dialed interface in a channel
datastore and checking the list before dialing out. This works, but currently
breaks situations like A calls B, A transfers B to C, B transfers C to A, and A
transfers C to B. Since human interaction is happening here and not an
automated forwarding loop, it should be allowed.
This patch removes the dialed_interfaces datastore when a call is bridged (a
suggestion from the brilliant mmichelson). If a call is being bridged, it
should be safe to assume that we aren't stuck in a loop.
Since we are now handling this is the bridge code, the previous attempts at
handling it in app_dial and app_queue are removed.
Review: https://reviewboard.asterisk.org/r/1195/
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r315645 | rmudgett | 2011-04-26 17:14:31 -0500 (Tue, 26 Apr 2011) | 21 lines
The 'e' special extension fails to trigger in at least two cases.
The 'e' extension is a fall back for the 'i', 't', or 'T' extensions if
any of them do not exist. Many of the places the 'e' extension was
supposed to be invoked fail because the priority was set wrong. There
were two places where the 'e' extension was not even checked for fall
back.
* Made invoke the 'e' extension similarly to the previous 'i', 't', or 'T'
extension check and added the 'e' extension as a fall back to the two
missing locations.
* Prioritized and optimized some hangup tests associated with the 'e'
extension.
(closes issue #19136)
Reported by: kshumard
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1196/
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r315503 | tilghman | 2011-04-26 14:32:50 -0500 (Tue, 26 Apr 2011) | 28 lines
Merged revisions 315502 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r315502 | tilghman | 2011-04-26 14:22:52 -0500 (Tue, 26 Apr 2011) | 21 lines
Merged revisions 315501 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r315501 | tilghman | 2011-04-26 14:18:46 -0500 (Tue, 26 Apr 2011) | 14 lines
Fix the bounds-checking code.
The code that set the bit within the select bitfield was correct, but the
bounds-checking code was not. The change to that line uses the new _bitsize
macro for clarity. Also, FD_ZERO macro did not zero-out anything but the
first word of the bitfield, so this could have caused problems with modules
using that macro with the expanded bitfield.
(closes issue #18773)
Reported by: jamicque
Patches:
20110423__issue18773.diff.txt uploaded by tilghman (license 14)
Tested by: chris-mac
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r315446 | russell | 2011-04-26 12:40:23 -0500 (Tue, 26 Apr 2011) | 14 lines
chan_local: resolve a deadlock.
This patch resolves a fairly complex deadlock that can occur with the
combination of chan_local and a dialplan switch, such as dynamic realtime
extensions, which pulls autoservice into the picture when doing a dialplan
lookup.
(closes issue #18818)
Reported by: nic
Patches:
issue18818.patch uploaded by jthurman (license 614)
18818.v1.txt uploaded by russell (license 2)
Tested by: nic, jthurman, kterzi, steve-howes, sysreq, IshMalik
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r314780 | russell | 2011-04-22 09:02:23 -0500 (Fri, 22 Apr 2011) | 18 lines
Merged revisions 314778 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r314778 | russell | 2011-04-22 08:58:03 -0500 (Fri, 22 Apr 2011) | 11 lines
Initialize buffers in getvar and getvarfull.
Initialize the buffers used to hold the result from GET VARIABLE or
GET VARIABLE FULL. The bug report shows func_read returning garbage in
the result. It assumed that the buffer passed in was initialized, like many
other functions do. In the more common code path (through the dialplan), it
is initialized, so just initialize it here too.
(closes issue #19050)
Reported by: johnz
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PRIShowSpans works like the AMI action DAHDIShowChannels but for PRI
spans. It is similar to the CLI command "pri show spans".
(closes issue #15980)
Reported by: dwery
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@314735 f38db490-d61c-443f-a65b-d21fe96a405b
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r314628 | mnicholson | 2011-04-21 13:24:05 -0500 (Thu, 21 Apr 2011) | 27 lines
Merged revisions 314620 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r314620 | mnicholson | 2011-04-21 13:22:19 -0500 (Thu, 21 Apr 2011) | 20 lines
Merged revisions 314607 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r314607 | mnicholson | 2011-04-21 13:19:21 -0500 (Thu, 21 Apr 2011) | 14 lines
Added limits to the number of unauthenticated sessions TCP based protocols are allowed to have open simultaneously. Also added timeouts for unauthenticated sessions where it made sense to do so.
Unrelated, the manager interface now properly checks if the user has the "system" privilege before executing shell commands via the Originate action.
AST-2011-005
AST-2011-006
(closes issue #18787)
Reported by: kobaz
(related to issue #18996)
Reported by: tzafrir
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r314550 | twilson | 2011-04-20 17:23:04 -0700 (Wed, 20 Apr 2011) | 13 lines
Merged revisions 314549 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r314549 | twilson | 2011-04-20 17:17:34 -0700 (Wed, 20 Apr 2011) | 6 lines
Don't allocate more space than necessary for a sip_pkt
This extra allocation is a hold-over from when pkt->data was a
character array. Now that it is an allocated string, just allocate
enough for the sip_pkt.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@314551 f38db490-d61c-443f-a65b-d21fe96a405b
Previously, the DAHDI format bit fields matched up with the Asterisk
bitfields. Since the Asterisk codec bit fields were replaced in r306010,
codec_dahdi needs to contain the formats itself. In the future, the DAHDI
formats should either change to something other than bitfields, or the
bitfields need to move from include/dahdi/kernel.h to
include/dahdi/user.h.
Signed-off-by: Shaun Ruffell <sruffell@digium.com>
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@314471 f38db490-d61c-443f-a65b-d21fe96a405b
The "controlling user number" is always the number of the voice mail box
which is identical with the subscriber number itself. This number which
is listed in the ISDN phone MWI menu cannot be called back to contact the
voice mail box. The controlling user number should be made configurable.
JIRA ABE-2738
JIRA SWP-2846
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r314069 | rmudgett | 2011-04-18 11:10:10 -0500 (Mon, 18 Apr 2011) | 22 lines
The AsyncAGI command loop is lax in the value it returns for the return status.
* Return correct status: SUCCESS/FAILED/HANGUP. Previously, abnormal
exits from the command loop such as hangup would return SUCCESS.
* The "asyncagi break" command now returns SUCCESS and is now the only way
to break the command loop with that status. Previously, it returned
FAILED.
* The AMI event AsyncAGI End is no longer sent if the AsyncAGI Start event
is not sent. Previously, this happened because of an error setting up the
AGI pipes.
* All executed AGI commands now get an AsyncAGI Exec result event.
Previously, if the command returned failure (because of hangup), the
command loop just exited with FAILURE and did not send the AsyncAGI Exec
result event.
* Makes sure that the channel frame queue is empty on hangup.
Review: https://reviewboard.asterisk.org/r/1183/
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r314067 | dvossel | 2011-04-18 10:23:45 -0500 (Mon, 18 Apr 2011) | 22 lines
Remove the need for deadlock avoidance in chan_sip do_monitor.
Deadlock avoidance between the sip pvt and the pvt->owner is
very difficult. Now that channel's are ao2 objects, this complication
is no longer necessary. It turns out the pvt's msg queue only
exists because of deadlock avoidance (when deadlock avoidance fails
msgs were added to a queue to be processed later), so this goes away as well.
The technique used in the new sip_lock_pvt_full() function should
be used as a template for replacing all locations where deadlock
avoidance occurs between a channel tech_pvt and the pvt's owner.
My hope is that this will begin a reversal of the invalid channel
driver locking architecture we have been using for so long.
This patch also resolves an issue where the pvt->owner gets
unlocked during processing the msg queue.
(closes issue #18690)
Reported by: dvossel
Review: https://reviewboard.asterisk.org/r/1182/
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r314017 | dvossel | 2011-04-18 08:41:06 -0500 (Mon, 18 Apr 2011) | 17 lines
sip codec negotiation of dynamic rtp payloads error fix
This patch fixes how chan_sip handles dynamic rtp payload types
it does not understand. At the moment if a dynamic payload's mime
type does not match one we understand, the payload does not get
removed from our payload table. As a result of this, the payload
is set to whatever dynamic codec we use internally for that payload
number on outgoing INVITES. This is incorrect.
This patch fixes this by properly checking the rtpmap set function's
return code to make sure it was found. The function can return both
-1 and -2 depending on the source of the mismatch. We were just
checking -1 explicitly.
Review: https://reviewboard.asterisk.org/r/1169/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@314018 f38db490-d61c-443f-a65b-d21fe96a405b
This fixes a regression in the media architecture change
where video frames did not have their video mark set
correctly. dvossel wrote this. twilson kindly committed
this, mmichelson found the bug.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@313822 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r313780 | rmudgett | 2011-04-14 15:59:56 -0500 (Thu, 14 Apr 2011) | 20 lines
Leftover debug messages unconditionally sent to the console.
Executing Dial(DAHDI/1/18475551212,300,) with the echotraining config
option enabled outputs the following debug messages unconditionally:
Dialing T1847555121 on 1
Dialing www2w on 1
* Made debug messages in my_dial_digits() normal debug messages that do
not get output unless enabled.
* Reworded some debug messages in my_dial_digits() to be clearer.
* Replace strncpy() with ast_copy_string() in my_dial_digits() which does
the same job better.
(closes issue #18847)
Reported by: vmikhelson
Tested by: rmudgett
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Add Asterisk Device State information and callbacks to the Call Completion
Supplemental Services for generic agents.
There are currently not many devices that have native support for CCSS.
Even as the devices become available there may be other reasons why one
may choose to not take advantage of the native abilities and stick with
the generic implementation. The generic implementation is quite capable
and could be greatly enhanced by adding device state capabilities. A
phone could then subscribe to the device state with a BLF key in
conjunction with Asterisk hints.
The advantages of the device state information would allow a single button
to: request CCSS, cancel a CCSS request, and display the current state of
a CCSS request.
For example, you may have a single button that when not lit, there is no
active CCSS request. When you press that button, the dialplan can query
the DEVICE_STATE() associated with that caller to determine whether they
should be calling CallCompletionRequest() or CallCompletionCancel(). If
there is currently a pending request, then the dialplan would cancel it.
This also has the advantage of showing the true state of a request, which
is an asynchronous call, even when CallCompletionRequest() thinks it was
successful. The actual request could ultimately fail. Once lit, further
feedback can be provided to the caller about the current state of their
request since it will be updated by the CCSS State Machine as appropriate.
The DEVICE_STATE mapping is configurable since the BLF being used on a
given phone type may vary. The idea is to allow some level of
customization as to the phone's behavior.
As an example, you may want the BLF key to go solid once you have
requested a callback. You may then want the LED to blink (typically
ringing) when either the callback is in process, which is a visual
indication that the incoming call is the desired callback. You may want
it to blink when the callee is ready but you are busy, giving you a visual
indication that the target is available as you may want to get off the
line so that the callback can be successful.
Device state information is sent back via the ast_devstate_prov_add()
callback for any generic CCSS device as it traverses through the state
machine. You simply provide a map between CC_STATE values and the
corresponding AST_DEVICE state values.
You could then generate hints against these states similar to what is
possible today with Custom Devstates or MeetMe states. For example, you
may have an extension 3000 that is currently associated with device
SIP/3000. You could then create a feature code for that extension that
may look something like:
exten => *823000,hint,ccss:sip/3000
You would then subscribe a BLF button to *823000 which would point to the
dialplan that handled CCSS requests/cancels using the available
DEVICE_STATE() information about ccss:sip/3000 to make the decision about
what to do.
(closes issue #18788)
Reported by: p_lindheimer
Patches:
ccss.trunk.18788.patch uploaded by p lindheimer (license 558)
Modified with final reviewboard comments.
Tested by: p_lindheimer, loloski
Review: https://reviewboard.asterisk.org/r/1105/
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r313700 | rmudgett | 2011-04-13 17:52:47 -0500 (Wed, 13 Apr 2011) | 5 lines
Revert flushing stale AsyncAGI commands from -r313615.
It looks like it was intentional to leave any commands or in-flight
commands in the queue in case Async AGI is run again on the call.
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r313588 | rmudgett | 2011-04-13 11:31:50 -0500 (Wed, 13 Apr 2011) | 55 lines
Merged revisions 313579 via svnmerge from
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r313579 | rmudgett | 2011-04-13 11:29:49 -0500 (Wed, 13 Apr 2011) | 48 lines
Merged revisions 313545 via svnmerge from
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r313545 | rmudgett | 2011-04-13 11:21:24 -0500 (Wed, 13 Apr 2011) | 41 lines
Asterisk does not hangup a channel after endpoint hangs up.
If the call that the dialplan started an AGI script for is hungup while
the AGI script is in the middle of a command then the AGI script is not
notified of the hangup. There are many AGI Exec commands that this can
happen with. The reported applications have been: Background, Wait, Read,
and Dial. Also the AGI Get Data command.
* Don't wait on the Asterisk channel after it has hung up. The channel is
likely to never need servicing again.
* Restored the AGI script's ability to return the AGI_RESULT_HANGUP value
in run_agi(). It previously only could return AGI_RESULT_SUCCESS or
AGI_RESULT_FAILURE after the DeadAGI and AGI applications were merged.
(closes issue #17954)
Reported by: mn3250
Patches:
issue17954_v1.8.patch uploaded by rmudgett (license 664)
issue17954_v1.6.2.patch uploaded by rmudgett (license 664)
issue17954_v1.4.patch uploaded by rmudgett (license 664)
Tested by: rmudgett
JIRA SWP-2171
(closes issue #18492)
Reported by: devmod
Tested by: rmudgett
JIRA SWP-2761
(closes issue #18935)
Reported by: nvitaly
Tested by: astmiv, rmudgett
JIRA SWP-3216
(closes issue #17393)
Reported by: siby
Tested by: rmudgett
JIRA SWP-2727
Review: https://reviewboard.asterisk.org/r/1165/
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r313517 | rmudgett | 2011-04-12 17:35:53 -0500 (Tue, 12 Apr 2011) | 12 lines
Bring the dumpchan application inline with "core show channel".
* Added fields that are in "core show channel" to dumpchan output.
* Fixed reuse of formatbuf before the previous string stored there was
used by snprintf. All output strings now have their own buffer.
* Adjusted the buffer sizes to not be so abusive of the stack now that
there are more buffers.
Change requested by oej.
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IPv6 support for ooh323,
bindaddr, peers and users ip can be IPv4 or IPv6 addr
correction for multi-homed mode (0.0.0.0 or :: bindaddr)
can work in dual 6/4 mode with :: bindaddr
gatekeeper mode isn't supported in v6 mode while
(issue #18278)
Reported by: may213
Patches:
ipv6-ooh323.patch uploaded by may213 (license 454)
Review: https://reviewboard.asterisk.org/r/1004/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@313482 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
also went ahead and fixed the problem it introduces before committing.
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r313435 | jrose | 2011-04-12 13:44:44 -0500 (Tue, 12 Apr 2011) | 1 line
fixing stupid mistake with putting code before variable declaration
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r313432 | jrose | 2011-04-12 13:12:29 -0500 (Tue, 12 Apr 2011) | 14 lines
reload Chan_dahdi memory leak caused by variables
chan_dahdi reloading with variables set via setvar in chan_dahdi.conf would
stay in the dahdi_pvt structs for individual channels (causing them to just
continue adding the new ones to the list) and also there was a memory leak
causes by the conf objects. This patch resolves both of these by using
ast_variables_destroy during the loading process.
(closes issue #17450)
Reported by: nahuelgreco
Patches:
patch.diff uploaded by jrose (license 1225)
Tested by: tilghman, jrose
Review: https://reviewboard.asterisk.org/r/1170/
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r313368 | rmudgett | 2011-04-11 18:03:02 -0500 (Mon, 11 Apr 2011) | 2 lines
Backport a restructuring change from trunk to make the next change stand out.
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r313369 | rmudgett | 2011-04-11 18:08:02 -0500 (Mon, 11 Apr 2011) | 13 lines
Frames from the inbound channel should go to all outbound channels in app_dial.c.
In app_dial.c:wait_for_answer() frames from the inbound channel should be
sent to all outbound channels instead of only if there is just one
outbound channel.
Control frames like AST_CONTROL_CONNECTED_LINE need to be passed to all of
the the outbound channels. This can happen if a blond transfer is done by
a remote switch on the inbound channel.
JIRA AST-443
JIRA SWP-2730
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r313190 | rmudgett | 2011-04-11 10:40:30 -0500 (Mon, 11 Apr 2011) | 39 lines
Merged revisions 313189 via svnmerge from
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r313189 | rmudgett | 2011-04-11 10:32:53 -0500 (Mon, 11 Apr 2011) | 32 lines
Merged revisions 313188 via svnmerge from
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r313188 | rmudgett | 2011-04-11 10:27:52 -0500 (Mon, 11 Apr 2011) | 25 lines
Stuck channel using FEATD_MF if caller hangs up at the right time.
The cause was actually a caller hanging up just at the end of the Feature
Group D DTMF tones that setup the call. The reason for this is a "guard
timer" that's implemented using ast_safe_sleep(100). If the caller
happens to hang up AFTER the final tone of the DTMF string but BEFORE the
end of that ast_safe_sleep(), then ast_safe_sleep() will return non-zero.
This causes the code to bounce to the end of ss_thread(), but it does NOT
tear down the call properly.
This should be a rare occurrence because the caller has to hang up at
EXACTLY the right time. Nonetheless, it was happening quite regularly on
the reporter's system. It's not easily reproducible, unless you purposely
increase the guard-time to 2000 or more. Once you do that, you can
reproduce it every time by watching the DTMF debug and hanging up just as
it ends.
Simply add an ast_hangup() before goto quit.
(closes issue #15671)
Reported by: jcromes
Patches:
issue15671.patch uploaded by pabelanger (license 224)
Tested by: jcromes
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r313001 | alecdavis | 2011-04-07 22:19:31 +1200 (Thu, 07 Apr 2011) | 13 lines
Fix ISDN calling subaddr User Specified Odd/Even Flag
Calculation of the Odd/Even flag was wrong.
Implement correct algo, and set odd/even=0 if data would be truncated.
Only allow automatic calculation of the O/E flag, don't let dialplan influence.
(closes issue #19062)
Reported by: festr
Patches:
bug19062.diff2.txt uploaded by alecdavis (license 585)
Tested by: festr, alecdavis, rmudgett
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r312949 | rmudgett | 2011-04-05 13:45:24 -0500 (Tue, 05 Apr 2011) | 6 lines
Crash if ISDN span layer 1 is down on initial load.
Regression from -r312575 B channel shifting during negotiation.
* Also combine updating the alarm flag with clearing the resetting flag.
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r312866 | rmudgett | 2011-04-05 10:38:14 -0500 (Tue, 05 Apr 2011) | 15 lines
Responding to OPTIONS packet with 404 because Asterisk not looking for "s" extension.
The get_destination() function was not using the "s" extension when the
request URI did not specify an extension. This is a regression caused
when the URI parsing code was extracted into parse_uri().
Made get_destination() substitute the "s" extension when the parsed URI
results in an empty string.
(closes issue #18348)
Reported by: shmaize
Patches:
issue18348_v1.8.patch uploaded by rmudgett (license 664)
Tested by: shmaize
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If the user invokes 'dialplan add extension' into a non-existing context, the context will be created
and a message informing the user of the context being created will be issued in cli.
(closes issue #17431)
Reported by: leearcher
Patches:
context_auto_create.diff uploaded by kobaz (license 834)
Tested by: leearcher, kobaz, jrose
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@312678 f38db490-d61c-443f-a65b-d21fe96a405b
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r312575 | rmudgett | 2011-04-04 11:10:50 -0500 (Mon, 04 Apr 2011) | 52 lines
Merged revisions 312574 via svnmerge from
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r312574 | rmudgett | 2011-04-04 11:00:02 -0500 (Mon, 04 Apr 2011) | 45 lines
Merged revisions 312573 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r312573 | rmudgett | 2011-04-04 10:49:30 -0500 (Mon, 04 Apr 2011) | 38 lines
Issues with ISDN calls changing B channels during call negotiations.
The handling of the PROCEEDING message was not using the correct call
structure if the B channel was changed. (The same for PROGRESS.) The call
was also not hungup if the new B channel is not provisioned or is busy.
* Made all call connection messages (SETUP_ACKNOWLEDGE, PROCEEDING,
PROGRESS, ALERTING, CONNECT, CONNECT_ACKNOWLEDGE) ensure that they are
using the correct structure and B channel. If there is any problem with
the operations then the call is now hungup with an appropriate cause code.
* Made miscellaneous messages (INFORMATION, FACILITY, NOTIFY) find the
correct structure by looking for the call and not using the channel ID.
NOTIFY is an exception with versions of libpri before v1.4.11 because a
call pointer is not available for Asterisk to use.
* Made all hangup messages (DISCONNECT, RELEASE, RELEASE_COMPLETE) find
the correct structure by looking for the call and not using the channel
ID.
(closes issue #18313)
Reported by: destiny6628
Tested by: rmudgett
JIRA SWP-2620
(closes issue #18231)
Reported by: destiny6628
Tested by: rmudgett
JIRA SWP-2924
(closes issue #18488)
Reported by: jpokorny
JIRA SWP-2929
JIRA AST-437 (The issues fixed here are most likely causing this JIRA issue.)
JIRA DAHDI-406
JIRA LIBPRI-33 (Stuck resetting flag likely fixed)
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r312509 | rmudgett | 2011-04-01 18:15:42 -0500 (Fri, 01 Apr 2011) | 22 lines
When a call going out an NT-PTMP port gets rejected, Asterisk crashes.
If a call is sent to an ISDN phone that rejects the call with
RELEASE_COMPLETE(cause: call reject(21), or busy(17)) Asterisk crashes.
I could not get my setup to crash. However, I could see the possibility
from a race condition between queuing an AST_CONTROL_BUSY to the core and
then queueing an AST_CONTROL_HANGUP. If the AST_CONTROL_BUSY is processed
before the AST_CONTROL_HANGUP is queued, the ast_channel could be
destroyed out from under chan_misdn.
Avoid this particular crash scenario by not queueing the
AST_CONTROL_HANGUP if the AST_CONTROL_BUSY was queued.
(closes issue #18408)
Reported by: wimpy
Patches:
issue18408_v1.8.patch uploaded by rmudgett (license 664)
Tested by: rmudgett, wimpy
JIRA SWP-2679
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r312461 | rmudgett | 2011-04-01 16:31:39 -0500 (Fri, 01 Apr 2011) | 25 lines
CallCompletionRequest()/CallCompletionCancel() exit non-zero if fail.
The CallCompletionRequest()/CallCompletionCancel() dialplan applications
exit nonzero on normal failure conditions. The nonzero exit causes the
dialplan to hangup immediately. The dialplan author has no opportunity to
report success/failure to the user.
* Made always return zero so the dialplan can continue.
* Made set CC_REQUEST_RESULT/CC_REQUEST_REASON and
CC_CANCEL_RESULT/CC_CANCEL_REASON channel variables respectively. Also
documented the values set.
* Reduced the warning about no core instance in CallCompletionCancel() to
a debug message. It is a normal event and should not be output at the
WARNING level.
(closes issue #18763)
Reported by: p_lindheimer
Patches:
ccss.patch uploaded by p lindheimer (license 558) Modified
Tested by: p_lindheimer, rmudgett
JIRA SWP-3042
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In chan_dahdi.conf, the user can now use length 4 patterns in addition to the usual length 2 patterns. The s
ntax remains the same and the method used to track the pattern history will only change when using the length
4 patterns.
(closes issue SWP-3250)
Code:
jrose
rmudgett
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r312211 | alecdavis | 2011-04-01 22:03:11 +1300 (Fri, 01 Apr 2011) | 36 lines
Merged revisions 312210 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r312210 | alecdavis | 2011-04-01 21:47:29 +1300 (Fri, 01 Apr 2011) | 29 lines
Merged revisions 312174 via svnmerge from
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r312174 | alecdavis | 2011-04-01 21:29:49 +1300 (Fri, 01 Apr 2011) | 23 lines
voicemail: get real last_message_index and count_messages, ODBC resequence
change last_message_index to read the max msgnum stored in the database
change count_messages to actually count the number of messages.
last_message_index change:
This fixed overwriting of the last message if msgnum=0 was missing.
Previously every incoming message would overwrite msgnum=1.
count_messages change:
allows us to detect when requencing is required in opneA_mailbox.
resequence enabled for ODBC storage:
Assists with fixing up corrupt databases with gaps, but only when
a user actively opens there mailboxes.
(closes issue #18692,#18582,#19032)
Reported by: elguero
Patches:
based on odbc_resequence_mailbox2.1.diff uploaded by elguero (license 37)
Tested by: elguero, nivek, alecdavis
Review: https://reviewboard.asterisk.org/r/1153/
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r312117 | alecdavis | 2011-04-01 20:32:12 +1300 (Fri, 01 Apr 2011) | 29 lines
Merged revisions 312103 via svnmerge from
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r312103 | alecdavis | 2011-04-01 20:25:54 +1300 (Fri, 01 Apr 2011) | 22 lines
Merged revisions 312070 via svnmerge from
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r312070 | alecdavis | 2011-04-01 19:46:56 +1300 (Fri, 01 Apr 2011) | 16 lines
app_voicemail: close_mailbox needs to respect additional messages while mailbox is open.
close_mailbox leave gaps in message sequence if messages are deleted and new messages
arrive during this time, this is because the shuffle down to slot 0, only shuffles
the number of pre-existing messages when mailbox is opened, ignoring new arrivals.
Fix: in close_mailbox re-evaluate number of messages before the shuffle, this then includes new arrivals.
Happens on filebased or ODBC storage.
(issues #19032,#18582,#18692,#18998)
Reported by: alecdavis,tootai,afosorio
Review: https://reviewboard.asterisk.org/r/1153/
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r312022 | rmudgett | 2011-03-31 15:11:40 -0500 (Thu, 31 Mar 2011) | 14 lines
chan_misdn segfaults when DEBUG_THREADS is enabled.
The segfault happens because jb->mutexjb is uninitialized from the
ast_malloc(). The internals of ast_mutex_init() were assuming a nonzero
value meant mutex tracking initialization had already happened. Recent
changes to mutex tracking code to reduce excessive memory consumption
exposed this uninitialized value.
Converted misdn_jb_init() to use ast_calloc() instead of ast_malloc().
Also eliminated redundant zero initialization code in the routine.
(closes issue #18975)
Reported by: irroot
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r311612 | bbryant | 2011-03-23 17:45:46 -0400 (Wed, 23 Mar 2011) | 9 lines
Fix a possible crash in sip/reqresp_parser.c that is caused by a possible null
value.
(closes issue #18821)
Reported by: cmaj
Patches:
patch-reqresp_parser_sip_uri_domain_cmp_c_locale-crash-1.8.3-rc2.diff.tx
uploaded by cmaj (license 830)
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r311558 | twilson | 2011-03-22 19:24:53 -0700 (Tue, 22 Mar 2011) | 5 lines
Don't use static declared buf in parse_name_andor_addr
This function isn't used anywhere yet, but we definitely don't want
to keep the same value for buf between calls to the function.
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dect_read() should not be called since audio is queued when available
through dect_dl_u_data_ind(). Return ast_null_frame so the core won't
hang up the channel and print a warning.
Signed-off-by: Patrick McHardy <kaber@trash.net>
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r311352 | jrose | 2011-03-18 11:19:05 -0500 (Fri, 18 Mar 2011) | 10 lines
Changes some print statements/events to use a blank string in place of NULL if the string in question is NULL.
This is supposed to improve Solaris compatibility since Solaris goes berserk when trying to output NULL strings.
(closes issue #18759)
Reported by: bklang
Patches:
null-strings.patch uploaded by bklang (license 919)
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r311297 | rmudgett | 2011-03-17 21:59:05 -0500 (Thu, 17 Mar 2011) | 12 lines
Race condition when ISDN CallRerouting/CallDeflection invoked.
The queued AST_CONTROL_BUSY could sometimes be processed before the
call_forward dial string is recognized.
* Moved setting the call_forwarding dial string after sending a response
to the initiator and just queue an empty frame to wake up the media thread
instead of an AST_CONTROL_BUSY.
* Added check for empty rerouting/deflection number and respond with an
error.
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r311295 | rmudgett | 2011-03-17 21:22:07 -0500 (Thu, 17 Mar 2011) | 35 lines
Merged revision 310986 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
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r310986 | rmudgett | 2011-03-16 13:56:28 -0500 (Wed, 16 Mar 2011) | 28 lines
Dial() o option broke when connected line feature added.
The patch restores the o option behavior and adds the ability to specify
the CallerID. The Dial o and f options are complementary to each other.
The o option stores the CallerID on the outgoing channel as the channel's
CallerID. The f option forces the CallerID sent by the outgoing channel.
o(x) - The argument 'x' is optional. If not present, then specify that
the CallerID that was present on the *calling* channel be stored as the
CallerID on the *called* channel. This was the behavior of Asterisk 1.0
and earlier. If present, then specify the CallerID stored on the *called*
channel. Note that o(${CALLERID(all)}) is similar to option o without
parameters.
f(x) - The argument 'x' is optional and its presence changes the behavior
of this option. If not present, then force the outgoing CallerID on a
call-forward or deflection to the dialplan extension for this Dial() using
a dialplan 'hint'. For example, some PSTNs do not allow CallerID to be
set to anything other than the numbers assigned to you. If present, then
force the outgoing CallerID to 'x'.
Patches:
jira_abe_2752_dial_fo_options.patch uploaded by rmudgett (license 664)
Tested by: rmudgett
JIRA ABE-2752
JIRA SWP-3096
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r311197 | jrose | 2011-03-17 14:03:34 -0500 (Thu, 17 Mar 2011) | 11 lines
This fixes a nasty chanspy bug which was causing a channel leak every time a spied on channel made a call.
In addition to the above, it makes certain channel destruction occurs so that applications don't get stuck waiting for datastore destruction while monitored by chanspy.
(closes issue #18742)
Reported by: jkister
Tested by: jkister, jcovert, jrose
Review: http://reviewboard.digium.internal/r/106/
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r310902 | twilson | 2011-03-16 12:19:57 -0500 (Wed, 16 Mar 2011) | 43 lines
Merged revisions 310889 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r310889 | twilson | 2011-03-16 12:03:27 -0500 (Wed, 16 Mar 2011) | 36 lines
Merged revisions 310888 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r310888 | twilson | 2011-03-16 11:58:42 -0500 (Wed, 16 Mar 2011) | 29 lines
Don't delay DTMF in core bridge while listening for DTMF features
This patch is mostly the work of Olle Johansson. I did some cleanup and
added the silence generating code if transmit_silence is set.
When a channel listens for DTMF in the core bridge, the outbound DTMF is not
sent until we have received DTMF_END. For a long DTMF, this is a disaster. We
send 4 seconds of DTMF to Asterisk, which sends no audio for those 4 seconds.
Some products see this delay and the time skew on RTP packets that results and
start ignoring the audio that is sent afterward.
With this change, the DTMF_BEGIN frame is inspected and checked. If it matches
a feature code, we wait for DTMF_END and activate the feature as before. If
transmit_silence=yes in asterisk.conf, silence is sent if we paritally match a
multi-digit feature. If it doesn't match a feature, the frame is forwarded
along with the DTMF_END without delay. By doing it this way, DTMF is not delayed.
(closes issue #15642)
Reported by: jasonshugart
Patches:
issue_15652_dtmf_ast-1.4.patch.txt uploaded by twilson (license 396)
Tested by: globalnetinc, jde
(closes issue #16625)
Reported by: sharvanek
Review: https://reviewboard.asterisk.org/r/1092/
Review: https://reviewboard.asterisk.org/r/1125/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@310941 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
(closes issue #18693)
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r310734 | may | 2011-03-15 00:45:53 +0300 (Tue, 15 Mar 2011) | 12 lines
Introduce t.38 parameters control functionality not full but enough for
Send/RcvFax support
Introduce t.38 controls between asterisk core and channel/proto layers.
Not all parameters are transferred from proto layers but *Fax apps
tested and work ok.
(issue #18693)
Reported by: benngard2
Patches:
issue-18693.patch uploaded by may213 (license 454)
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r310636 | rmudgett | 2011-03-14 11:50:59 -0500 (Mon, 14 Mar 2011) | 39 lines
Merged revisions 310635 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r310635 | rmudgett | 2011-03-14 11:47:54 -0500 (Mon, 14 Mar 2011) | 32 lines
Merged revisions 310633 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r310633 | rmudgett | 2011-03-14 11:38:24 -0500 (Mon, 14 Mar 2011) | 25 lines
"Caller*ID failed checksum" on Wildcard TDM2400P and TDM410
The last character in the caller id message is getting a framing error.
The checksum is the last character in the message. A framing error in the
checksum could be because:
1) The sender did not send a full stop bit.
2) The sender cut off the FSK carrier too soon.
3) The sender opted to send zero of the specified zero to 10 trailing mark
bits and round-off errors in the code resulted in the code not being where
it thought it was in the demodulated bit stream.
Bit 8 of 'b' is set when parity error.
Bit 9 of 'b' is set when framing error.
Made ignore the framing and parity error bits if the errored character is
the checksum. We can tolerate a framing/parity error there. The checksum
character validates the message.
(closes issue #18474)
Reported by: nivek
Patches:
callerid.c.1.patch uploaded by nivek (license 636) (with modifications)
Tested by: nivek
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r310462 | tilghman | 2011-03-12 14:27:54 -0600 (Sat, 12 Mar 2011) | 45 lines
Merged revisions 310448 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r310448 | tilghman | 2011-03-12 14:24:54 -0600 (Sat, 12 Mar 2011) | 38 lines
Recorded merge of revisions 310435 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r310435 | tilghman | 2011-03-12 14:22:07 -0600 (Sat, 12 Mar 2011) | 31 lines
Add AELSub, which provides a stable entry point into AEL subroutines.
This commit needs some explanation, given that we're adding a new application
into an existing release branch. This is generally a violation of our release
policy, except in very limited circumstances, and I believe this is one of
those circumstances.
The problem that this solves is one of the sanity of using multiple dialplan
languages to define a dialplan. In the case of the reporter, he or she is
using AEL is define subroutines, while using Realtime extensions to invoke
those subroutines. While you can do this, it's based upon the reality of AEL
using actual dialplan extensions; however, there is no guarantee that the
details of _how_ AEL is compiled into extensions will remain stable. In fact,
at the time of this commit, it has already changed twice, once in a
fundamental way.
Now normally, a new application would only be added to trunk. However, this
application is explicitly to create a stable user-level API between versions,
and adding it to trunk only will not solve the user's problem of switching
between 1.6.2 and 1.8, nor will it help anybody switching from 1.8 to 1.10.
Therefore, it needs to go into existing release branches. For the sake of
consistency, and also because one of the changes was between 1.4 and 1.6.x,
I am also electing to commit this to 1.4.
(closes issue #18910)
Reported by: alexandrekeller
Patches:
20110304__issue18919__1.6.2.diff.txt uploaded by tilghman (license 14)
20110304__issue18919__1.4.diff.txt uploaded by tilghman (license 14)
Tested by: alexandrekeller
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Recent versions of GCC have a tuning option value of 'native', which causes
the compiler to optimize the build for the CPU the compile is performed on.
Since most people are building Asterisk on the machine they plan to run it on,
the configure script and build system will now use this value unless a different
value is specified by the user in CFLAGS when the configure script is executed.
In addition, this value will be used for building the GSM and LPC10 codecs as
well, in preference to the logic that has been in their Makefiles forever to
optimize for certain types of CPUs.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@310332 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r310287 | alecdavis | 2011-03-11 19:47:44 +1300 (Fri, 11 Mar 2011) | 17 lines
remote_bridge_loop: prevent segfault when after transfer of IAX2 of DAHDI call
If the channel condition is one of the following after breaking out of the loop, don't try to update_peer
(where x = 0/1)
1). ZOMBIE
2). cx->tech_pvt != pvtx
3). gluex != ast_rtp_instance_get_glue(cx->tech->type))
(closes issue #18781)
Reported by: alecdavis
Patches:
bug18781.diff3.txt uploaded by alecdavis (license 585)
Tested by: alecdavis, ZX81
Review: https://reviewboard.asterisk.org/r/1128/
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r310240 | twilson | 2011-03-10 10:05:45 -0600 (Thu, 10 Mar 2011) | 13 lines
Add \r\n to remaining http headers passed to ast_http_send
r309204 changed the behavior of ast_http_send. It now requires headers
to be passed with trailing \r\n. This change updates the remaining
instances in the code that did not pass the \r\n.
(closes issue #18186)
Reported by: nivaldomjunior
Patches:
res_phoneprov.c.diff uploaded by lathama (license 1028)
manager.diff.txt uploaded by twilson (license 396)
Tested by: lathama
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r309808 | tilghman | 2011-03-06 18:54:42 -0600 (Sun, 06 Mar 2011) | 14 lines
Merged revisions 309251 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r309251 | tilghman | 2011-03-01 19:06:02 -0600 (Tue, 01 Mar 2011) | 7 lines
Revert previous 2 commits, and instead conditionally redefine the same macro used in flex 2.5.35 that clashed with our workaround.
Not surprisingly, the workaround was exactly the same code as was provided by
the Flex maintainers, albeit in two different places, in different macros.
This should fix the FreeBSD builds, which have an older version of Flex.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@309809 f38db490-d61c-443f-a65b-d21fe96a405b
Adding the setvar option with variable substitution on the value allows things
like setting the outbound caller id name to the summary of a calendar event,
etc. Values could be chained together as they are appended in order to do some
scripting if necessary.
Review: https://reviewboard.asterisk.org/r/1134/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@309640 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r309542 | mnicholson | 2011-03-04 13:00:33 -0600 (Fri, 04 Mar 2011) | 11 lines
Merged revisions 309541 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r309541 | mnicholson | 2011-03-04 12:59:20 -0600 (Fri, 04 Mar 2011) | 4 lines
Check for errors from fseek() when loading config file, properly abort on errors from fread(), and supply a traceback for errors generated when loading the config file.
Also, prepend a newline to traceback output so that the main error message is on it's own line.
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r309445 | rmudgett | 2011-03-04 09:22:04 -0600 (Fri, 04 Mar 2011) | 46 lines
Get real channel of a DAHDI call.
Starting with Asterisk v1.8, the DAHDI channel name format was changed for
ISDN calls to: DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number>
There were several reasons that the channel name had to change.
1) Call completion requires a device state for ISDN phones. The generic
device state uses the channel name.
2) Calls do not necessarily have B channels. Calls placed on hold by an
ISDN phone do not have B channels.
3) The B channel a call initially requests may not be the B channel the
call ultimately uses. Changes to the internal implementation of the
Asterisk master channel list caused deadlock problems for chan_dahdi if it
needed to change the channel name. Chan_dahdi no longer changes the
channel name.
4) DTMF attended transfers now work with ISDN phones because the channel
name is "dialable" like the chan_sip channel names.
For various reasons, some people need to know which B channel a DAHDI call
is using.
* Added CHANNEL(dahdi_span), CHANNEL(dahdi_channel), and
CHANNEL(dahdi_type) so the dialplan can determine the B channel currently
in use by the channel. Use CHANNEL(no_media_path) to determine if the
channel even has a B channel.
* Added AMI event DAHDIChannel to associate a DAHDI channel with an
Asterisk channel so AMI applications can passively determine the B channel
currently in use. Calls with "no-media" as the DAHDIChannel do not have
an associated B channel. No-media calls are either on hold or
call-waiting.
(closes issue #17683)
Reported by: mrwho
Tested by: rmudgett
(closes issue #18603)
Reported by: arjankroon
Patches:
issue17683_18603_v1.8_v2.patch uploaded by rmudgett (license 664)
Tested by: stever28, rmudgett
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r309256 | qwell | 2011-03-02 13:54:20 -0600 (Wed, 02 Mar 2011) | 15 lines
Merged revisions 309255 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r309255 | qwell | 2011-03-02 13:53:47 -0600 (Wed, 02 Mar 2011) | 8 lines
Fix usage of "hasvoicemail=yes" and "mailbox=" in users.conf for SIP.
Since it's a duplicate, nothing is going to be done, so delme doesn't need to
be set at all. Strangely, when this was added, this was being set to 1 in 1.6,
and 0 in trunk.
(issue AST-439)
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r309035 | tilghman | 2011-02-28 05:10:28 -0600 (Mon, 28 Feb 2011) | 15 lines
Merged revisions 309033-309034 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r309033 | tilghman | 2011-02-28 04:43:12 -0600 (Mon, 28 Feb 2011) | 4 lines
A later version of flex already includes the fwrite workaround code, which if used twice causes a compilation error.
Detect whether Flex will compile without the workaround; if so, suppress our workaround code.
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r309034 | tilghman | 2011-02-28 05:07:52 -0600 (Mon, 28 Feb 2011) | 2 lines
Clarify meaning, removing double negative (stupid!)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@309036 f38db490-d61c-443f-a65b-d21fe96a405b
Fix timers not firing and events not triggering. The exact reason is unknown,
synchronizing the event handler to chan_sip fixes it however.
Signed-off-by: Patrick McHardy <kaber@trash.net>
The kernel doesn't depend on getting correctly sized frames anymore, use
the frame size given by asterisk.
Signed-off-by: Patrick McHardy <kaber@trash.net>
'c' is only set if there is a RING_PATTERN environment variable,
'pattern' is the correct pointer to use.
Signed-off-by: Patrick McHardy <kaber@trash.net>
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r308945 | alecdavis | 2011-02-26 07:52:53 +1300 (Sat, 26 Feb 2011) | 21 lines
Fix Deadlock with attended transfer of SIP call
Call path
sip_set_rtp_peer (locks chan then pvt)
transmit_reinvite_with_sdp
try_suggested_sip_codec
pbx_builtin_getvar_helper (locks p->owner)
But by the time p->owner lock was attempted, seems as though chan and p->owner were different.
So in sip_set_rtp_peer, lock pvt first then lock p->owner using deadlocking methods.
(closes issue #18837)
Reported by: alecdavis
Patches:
bug18837-trunk.diff3.txt uploaded by alecdavis (license 585)
Tested by: alecdavis, Irontec, ZX81, cmaj
Review: [https://reviewboard.asterisk.org/r/1126/]
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r308903 | rmudgett | 2011-02-24 15:38:41 -0600 (Thu, 24 Feb 2011) | 9 lines
Invalid read in ast_channel_set_caller_event().
Valgrind reported that ast_channel_set_caller_event() was reading data
from a freed buffer when using the pre_set structure.
Rearange things to pre-calculate the name and number pointer before
updating the caller party structure to see if the name or number was
changed.
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r308815 | twilson | 2011-02-24 11:57:18 -0600 (Thu, 24 Feb 2011) | 26 lines
Merged revisions 308814 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r308814 | twilson | 2011-02-24 11:54:49 -0600 (Thu, 24 Feb 2011) | 19 lines
Merged revisions 308813 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r308813 | twilson | 2011-02-24 11:42:16 -0600 (Thu, 24 Feb 2011) | 12 lines
Don't broadcast FullyBooted to every AMI connection
The FullyBooted event should not be sent to every AMI connection every
time someone connects via AMI. It should only be sent to the user who
just connected.
(closes issue #18168)
Reported by: FeyFre
Patches:
bug0018168.patch uploaded by FeyFre (license 1142)
Tested by: FeyFre, twilson
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-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@308582 f38db490-d61c-443f-a65b-d21fe96a405b
Guessed the log levels based on info that level 3
is the soft roof. Can we create a page / document
to define the levels?
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@308527 f38db490-d61c-443f-a65b-d21fe96a405b
enable reporting of the request URI / URL in debugging
change funny debug note to a serious note.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@308372 f38db490-d61c-443f-a65b-d21fe96a405b
Modern browsers are checking for the MIME Type of pages
and in some cases will not load a file if the type is
wrong.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@308331 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r308010 | qwell | 2011-02-15 17:34:03 -0600 (Tue, 15 Feb 2011) | 24 lines
Merged revisions 308007 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r308007 | qwell | 2011-02-15 17:33:24 -0600 (Tue, 15 Feb 2011) | 17 lines
Merged revisions 308002 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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List the current mapping of DAHDI B channels to Asterisk channel names and
which calls are on hold or call-waiting. Calls on hold or call-waiting
are not associated with any B channel.
JIRA LIBPRI-27
JIRA SWP-2547
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r307879 | rmudgett | 2011-02-15 10:13:55 -0600 (Tue, 15 Feb 2011) | 37 lines
No response sent for SIP CC subscribe/resubscribe request.
Asterisk does not send a response if we try to subscribe for call
completion after we have received a 180 Ringing. You can only subscribe
for call completion when the call has been cleared.
When we receive the 180 Ringing, for this call, its call-completion state
is 'CC_AVAILABLE'. If we then send a subscribe message to Asterisk, it
trys to change the call-completion state to 'CC_CALLER_REQUESTED'.
Because this is an invalid state change, it just ignores the message. The
only state Asterisk will accept our subscribe message is in the
'CC_CALLER_OFFERED' state.
Asterisk will go into the 'CC_CALLER_OFFERED' when the SIP client clears
the call by sending a CANCEL.
Asterisk should always send a response. Even if its a negative one.
The fix is to allow for the CCSS core to notify a CC agent that a failure
has occurred when CC is requested. The "ack" callback is replaced with a
"respond" callback. The "respond" callback has a parameter indicating
either a successful response or a specific type of failure that may need
to be communicated to the requester.
(closes issue #18336)
Reported by: GeorgeKonopacki
Tested by: mmichelson, rmudgett
JIRA SWP-2633
(closes issue #18337)
Reported by: GeorgeKonopacki
Tested by: mmichelson
JIRA SWP-2634
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r307750 | tilghman | 2011-02-14 00:50:23 -0600 (Mon, 14 Feb 2011) | 23 lines
Calling a gosub routine defined in AEL from Dial/Queue ceased to work.
A bug in AEL did not distinguish between the "s" extension generated by
AEL and an "s" extension that was required to exist by the chan_dahdi
(or another channel) that was not supplied with a starting extension.
Therefore, AEL made incorrect assumptions about what commands were
permissable in the context. This was fixed by making AEL generate a
different extension name. However, Dial and Queue make additional
assumptions about the name of the default gosub extension. Therefore,
they needed to be brought into line with a "macro" rendered by AEL (as
a gosub), without breaking traditional dialplans written without the
aid of AEL.
Related to (issue #18480)
Reported by: nivek
(closes issue #18729)
Reported by: kkm
Patches:
20110209__issue18729.diff.txt uploaded by tilghman (license 14)
018729-dial-queue-gosub-try3.patch uploaded by kkm (license 888)
Tested by: kkm
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r307536 | qwell | 2011-02-10 16:39:30 -0600 (Thu, 10 Feb 2011) | 22 lines
Merged revisions 307535 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r307535 | qwell | 2011-02-10 16:35:49 -0600 (Thu, 10 Feb 2011) | 15 lines
Merged revisions 307534 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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The nativeformats field was being overwritten when it should have been
appended too. This caused some format capabilities to be lost briefly and
some log warnings to be output.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@307433 f38db490-d61c-443f-a65b-d21fe96a405b
small fixes.
Interpret remote side H.225 version.
Corrections for H.323v2 endpoints:
don't start TCS and MSD before connect,
don't start TCS and MSD by accepting H.245 connection,
start TCS and MSD by StartH245 facility message.
Other fixes:
fix non zeroended remoteDisplayName issue, small fixes in call clearing
by closing H.245 connection, tcp keepalive introduced on TCP
connections (now is hardcoded, will be configurable in the future),
don't force H.245tunneling if FastStart is active, don't send Alerting
singal more than once per call.
(closes issue #18542)
Reported by: vmikhelson
Patches:
issue18542-final-3.patch uploaded by may213 (license 454)
Tested by: vmikhelson
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@307396 f38db490-d61c-443f-a65b-d21fe96a405b
From the submitter:
I've added a new manager action to list only the active conferences on an
Asterisk system. It shows the same data displayed when you run a 'meetme list'
on the Asterisk CLI.
(closes issue #17905)
Reported by: rcasas
Patches:
app_meetme.c.patch uploaded by rcasas (license 641)
Review: https://reviewboard.asterisk.org/r/874/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@307359 f38db490-d61c-443f-a65b-d21fe96a405b
Clarify warning message when LOADABLE_MODULES is disabled but we still
try to load a module.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@307192 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r307092 | qwell | 2011-02-08 15:24:01 -0600 (Tue, 08 Feb 2011) | 9 lines
Fix issue with verbose messages not showing on remote console.
This code was reworked recently, and since the logchannel list hadn't been
created yet at this point, and it was a verbose message, it was being dropped
on the floor. Now it'll continue on to where it should be handled.
(closes issue #18580)
Reported by: pabelanger
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r307065 | mmichelson | 2011-02-08 15:13:08 -0600 (Tue, 08 Feb 2011) | 6 lines
Add a couple of useful channel variables for the CC recall macro.
CC_EXTEN and CC_CONTEXT will allow you to determine the channel
and context that will be called when the recall occurs.
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r306962 | jpeeler | 2011-02-08 13:25:38 -0600 (Tue, 08 Feb 2011) | 22 lines
Merged revisions 306961 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r306961 | jpeeler | 2011-02-08 13:25:10 -0600 (Tue, 08 Feb 2011) | 15 lines
Merged revisions 306960 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@306963 f38db490-d61c-443f-a65b-d21fe96a405b
Pass a MCID request to the bridged channel so the bridged channel can send
it to the network.
The ability to send the MCID request on an ISDN span is enabled with the
new chan_dahdi.conf mcid_send option.
JIRA SWP-2845
JIRA ABE-2736
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@306755 f38db490-d61c-443f-a65b-d21fe96a405b
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r306674 | twilson | 2011-02-07 14:43:22 -0800 (Mon, 07 Feb 2011) | 24 lines
Merged revisions 306673 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r306673 | twilson | 2011-02-07 14:40:20 -0800 (Mon, 07 Feb 2011) | 17 lines
Merged revisions 306672 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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/
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r306619 | twilson | 2011-02-07 14:15:27 -0800 (Mon, 07 Feb 2011) | 24 lines
Merged revisions 306618 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r306618 | twilson | 2011-02-07 13:59:54 -0800 (Mon, 07 Feb 2011) | 17 lines
Merged revisions 306617 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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/
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r306575 | mmichelson | 2011-02-07 11:36:56 -0600 (Mon, 07 Feb 2011) | 9 lines
Rearrange a bit of code in the generic CC recall operation.
By waiting to call the callback macro after the CC_INTERFACES,
extension, priority, and context have been set, this information
can be accessed more easily within the callback macro.
Reported by Philippe Lindheimer.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@306576 f38db490-d61c-443f-a65b-d21fe96a405b
The logic got reversed, oops. Works properly now when multiple blackfilters are
present.
(closes issue #18283)
Reported by: telecos82
Patches:
ast_managereventfilter.patch uploaded by telecos82 (license 687)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@306432 f38db490-d61c-443f-a65b-d21fe96a405b
The display ie handling can be controlled independently in the send and
receive directions with the following options:
* Block display text data.
* Use display text in SETUP/CONNECT messages for name.
* Use display text for COLP name updates (FACILITY/NOTIFY as appropriate).
* Pass arbitrary display text during a call. Sent in INFORMATION
messages. Received from any message that the display text was not used as
a name.
If the display options are not set then the options default to legacy
behavior.
The arbitrary display text is exchanged between bridged channels using the
AST_FRAME_TEXT frame type.
To send display text from the dialplan use the SendText() application when
the arbitrary display text option is enabled.
JIRA SWP-2688
JIRA ABE-2693
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@306396 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r306324 | rmudgett | 2011-02-04 12:53:06 -0600 (Fri, 04 Feb 2011) | 9 lines
Don't send redirecting updates to the caller if the dialplan forked the call.
Each fork in the dial could be redirected and confuse the caller. For
ISDN the DivLeg1 and DivLeg3 messages would get confused because ISDN
redirects calls in sequence not in parallel.
* Also fixed a formatting inconsistency in app_dial.c and make a warning
message more useful about what frame type could not be written.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@306325 f38db490-d61c-443f-a65b-d21fe96a405b
It seems extconf.c already defines some local ast_debug() functions. Theses
should be removed and replaced with logger.h. A patch will be added to
reviewboard shortly.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@306292 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r306215 | jpeeler | 2011-02-03 17:49:28 -0600 (Thu, 03 Feb 2011) | 20 lines
Fix SIP deadlock involving state changes.
Once again a call to pbx_builtin_getvar_helper (and pbx_builtin_setvar_helper)
has caused locking problems. Both of these functions lock the channel when
the channel argument is passed in!
In this case, the suspected problem (the backtrace makes it impossible to tell)
was the private being locked in sip_set_rtp_peer and then:
transmit_reinvite_with_sdp
try_suggested_sip_codec
pbx_builtin_getvar_helper
(Traced to verify that the fix was only required in 1.8 and later.)
(closes issue #18491)
Reported by: cmaj
Patches:
chan_sip_fix_deadlocks_bug_18491.txt uploaded by cmaj (license 830)
Tested by: cmaj
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r306127 | twilson | 2011-02-03 13:03:26 -0800 (Thu, 03 Feb 2011) | 23 lines
Merged revisions 306126 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r306126 | twilson | 2011-02-03 12:56:00 -0800 (Thu, 03 Feb 2011) | 16 lines
Merged revisions 306119 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
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r306124 | jpeeler | 2011-02-03 14:50:48 -0600 (Thu, 03 Feb 2011) | 17 lines
Merged revisions 306123 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r306123 | jpeeler | 2011-02-03 14:49:48 -0600 (Thu, 03 Feb 2011) | 10 lines
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)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@306125 f38db490-d61c-443f-a65b-d21fe96a405b
This patch is the foundation of an entire new way of looking at media in Asterisk.
The code present in this patch is everything required to complete phase1 of my
Media Architecture proposal. For more information about this project visit the link below.
https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal
The primary function of this patch is to convert all the usages of format
bitfields in Asterisk to use the new format and format_cap APIs. Functionally
no change in behavior should be present in this patch. Thanks to twilson
and russell for all the time they spent reviewing these changes.
Review: https://reviewboard.asterisk.org/r/1083/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@306010 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r305923 | rmudgett | 2011-02-02 18:24:40 -0600 (Wed, 02 Feb 2011) | 24 lines
Merged revisions 305889 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r305889 | rmudgett | 2011-02-02 18:15:07 -0600 (Wed, 02 Feb 2011) | 17 lines
Merged revisions 305888 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@305939 f38db490-d61c-443f-a65b-d21fe96a405b
sip show settings reports qualifyfreq in milliseconds.
sip.conf configures qualifyfreg in seconds.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@305650 f38db490-d61c-443f-a65b-d21fe96a405b
Update README, CHANGES, and Makefile. Direct users to
http://wiki.asterisk.org for documentation or to the
AST.txt and AST.pdf included in the tarball.
(closes issue #18443)
Reported by: bas
Patches:
changes.diff uploaded by lathama (license 1028)
readme.diff uploaded by lathama (license 1028)
Tested by: lathama bas
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@305561 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r305343 | rmudgett | 2011-01-31 18:01:09 -0600 (Mon, 31 Jan 2011) | 21 lines
Merged revisions 305342 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r305342 | rmudgett | 2011-01-31 17:50:10 -0600 (Mon, 31 Jan 2011) | 14 lines
Merged revisions 305341 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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r305254 | qwell | 2011-01-31 17:07:00 -0600 (Mon, 31 Jan 2011) | 24 lines
Merged revisions 305253 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r305253 | qwell | 2011-01-31 16:59:34 -0600 (Mon, 31 Jan 2011) | 17 lines
Merged revisions 305252 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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r305247 | qwell | 2011-01-31 16:25:23 -0600 (Mon, 31 Jan 2011) | 7 lines
Add alternative name for config option.
The SIP sample configuration had "tlscadir" as the option name, but chan_sip
used the more correct "tlscapath". Now both are accepted.
Discovered (sort of) by a user on IRC in #asterisk
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r304950 | tilghman | 2011-01-31 00:41:36 -0600 (Mon, 31 Jan 2011) | 18 lines
Change mutex tracking so that it only consumes memory in the core mutex object when it's actually being used.
This reduces the overall size of a mutex which was 3016 bytes before this back
down to 216 bytes (this is on 64-bit Linux with a glibc-implemented mutex).
The exactness of the numbers here may vary slightly based upon how mutexes are
implemented on a platform, but the long and short of it is that prior to this
commit, chan_iax2 held down 98MB of memory on a 64-bit system for nothing more
than a table of 32767 locks. After this commit, the same table occupies a mere
7MB of memory.
(closes issue #18194)
Reported by: job
Patches:
20110124__issue18194.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
Review: https://reviewboard.asterisk.org/r/1066
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@304951 f38db490-d61c-443f-a65b-d21fe96a405b
Add and extend the see-also sections to the documentation for applications
and functions in an effort to expand the online documentation of the wiki.
Also check for and update any links to moved documentation in the doc folder.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@304913 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r304774 | seanbright | 2011-01-29 12:54:43 -0500 (Sat, 29 Jan 2011) | 16 lines
Merged revisions 304773 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r304773 | seanbright | 2011-01-29 12:51:28 -0500 (Sat, 29 Jan 2011) | 9 lines
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
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r304730 | seanbright | 2011-01-29 12:15:27 -0500 (Sat, 29 Jan 2011) | 22 lines
Merged revisions 304729 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r304729 | seanbright | 2011-01-29 12:01:51 -0500 (Sat, 29 Jan 2011) | 15 lines
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
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r304727 | seanbright | 2011-01-29 11:28:27 -0500 (Sat, 29 Jan 2011) | 16 lines
Merged revisions 304726 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r304726 | seanbright | 2011-01-29 11:26:57 -0500 (Sat, 29 Jan 2011) | 9 lines
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
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r304683 | seanbright | 2011-01-28 17:54:23 -0500 (Fri, 28 Jan 2011) | 16 lines
Merged revisions 304659,304682 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r304659 | seanbright | 2011-01-28 16:22:09 -0500 (Fri, 28 Jan 2011) | 5 lines
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.
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r304682 | seanbright | 2011-01-28 17:38:05 -0500 (Fri, 28 Jan 2011) | 2 lines
Revert part of the previous commit that snuck in.
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r304638 | seanbright | 2011-01-28 15:19:08 -0500 (Fri, 28 Jan 2011) | 11 lines
Restore some conditionals that we lost in r277814.
There are some cases where ast_append_ha() is called with a NULL instead of a
valid int pointer. So if we get a NULL, don't try to dereference it.
(closes issue #18162)
Reported by: imcdona
Patches:
issue0018162.patch uploaded by pabelanger (license 224)
Tested by: enegaard
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@304639 f38db490-d61c-443f-a65b-d21fe96a405b
table name, because of the use of an uninitialized variable. Fixes an error
introduced in r300882.
(closes issue #18605)
Reported by: romain_proformatique
Patches:
res_config_pgsql_fix.patch uploaded by romain proformatique (license 975)
Tested by: romain_proformatique
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@304600 f38db490-d61c-443f-a65b-d21fe96a405b
Skipping the call to set_t38_fax_caps() caused the FAX session
details to not be marked as supporting audio FAX either... the
function's name is a bit misleading. This patch restores the
single bit of non-T.38 behavior from that function when audio
mode is forced.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@304599 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r304554 | rmudgett | 2011-01-27 13:08:14 -0600 (Thu, 27 Jan 2011) | 4 lines
Warning message if CALLCOMPLETION(cc_callback_macro or cc_agent_dialstring) are empty.
Test if the value pointer is not NULL instead of not ast_strlen_zero().
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r304466 | qwell | 2011-01-27 11:03:01 -0600 (Thu, 27 Jan 2011) | 23 lines
Merged revisions 304465 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r304465 | qwell | 2011-01-27 11:01:24 -0600 (Thu, 27 Jan 2011) | 16 lines
Merged revisions 304464 via svnmerge from
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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
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The recently added option to disable usage of T.38 for a single
session should have been named 'F' for 'force audio', since that
is really what the user is asking to happen (and it's a positive
option instead of a negative option that way).
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@304422 f38db490-d61c-443f-a65b-d21fe96a405b
Followme answers an incoming call if it hasn't already been answered and starts
MOH. Some poorly designed autodialers see the answer and start playing their
message to the hold music. The 'N' option has been added to indicate ringing and
not answer until the call is accepted.
(closes issue #18479)
Reported by: ianc
Patches:
trunk_followme.diff uploaded by ianc (license 998)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@304384 f38db490-d61c-443f-a65b-d21fe96a405b
Sometimes during troubleshooting it can be useful to disable T.38 usage in order
to narrow down a problem. This patch adds an 'n' option to SendFAX and ReceiveFAX
so that can be done without having to disable T.38 usage entirely for the peer
that Asterisk is communicating with.
(inspired by trying to assist Bryant Zimmerman on asterisk-users)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@304342 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r304097 | seanbright | 2011-01-25 20:26:26 -0500 (Tue, 25 Jan 2011) | 19 lines
Merged revisions 304096 via svnmerge from
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r304096 | seanbright | 2011-01-25 20:24:58 -0500 (Tue, 25 Jan 2011) | 12 lines
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
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r304007 | rmudgett | 2011-01-25 17:28:25 -0600 (Tue, 25 Jan 2011) | 22 lines
Merged revisions 304006 via svnmerge from
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r304006 | rmudgett | 2011-01-25 17:25:32 -0600 (Tue, 25 Jan 2011) | 15 lines
Merged revisions 304005 via svnmerge from
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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.
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r303962 | twilson | 2011-01-25 16:09:01 -0600 (Tue, 25 Jan 2011) | 30 lines
Merged revisions 303960 via svnmerge from
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r303960 | twilson | 2011-01-25 16:02:42 -0600 (Tue, 25 Jan 2011) | 23 lines
Merged revisions 303906 via svnmerge from
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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/
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r303771 | rmudgett | 2011-01-25 11:49:20 -0600 (Tue, 25 Jan 2011) | 54 lines
Merged revisions 303769 via svnmerge from
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r303769 | rmudgett | 2011-01-25 11:42:42 -0600 (Tue, 25 Jan 2011) | 47 lines
Merged revisions 303765 via svnmerge from
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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
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r303678 | jpeeler | 2011-01-25 11:02:38 -0600 (Tue, 25 Jan 2011) | 33 lines
Merged revisions 303677 via svnmerge from
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r303677 | jpeeler | 2011-01-25 10:59:28 -0600 (Tue, 25 Jan 2011) | 26 lines
Merged revisions 303676 via svnmerge from
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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)
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r303549 | russell | 2011-01-24 14:51:37 -0600 (Mon, 24 Jan 2011) | 45 lines
Merged revisions 303548 via svnmerge from
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r303548 | russell | 2011-01-24 14:49:53 -0600 (Mon, 24 Jan 2011) | 38 lines
Merged revisions 303546 via svnmerge from
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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/
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For each component, the set of valid BNF expansions defines exactly
which characters may appear unescaped. All other characters MUST be
escaped.
This patch modifies ast_uri_encode() to encode strings in line with this recommendation. This patch also adds an ast_escape_quoted() function which escapes '"' and '\' characters in quoted strings in accordance with section 25.1 of RFC 3261. The ast_uri_encode() function has also been modified to take an ast_flags struct describing the set of rules it should use when escaping characters to allow for it to escape SIP URIs in addition to HTTP URIs and other types of URIs or variations of those two URI types in the future.
The ast_uri_decode() function has also been modified to accept an ast_flags struct describing the set of rules to use when decoding to enable decoding '+' as ' ' in legacy http URLs.
The unit tests for these functions have also been updated.
ABE-2705
Review: https://reviewboard.asterisk.org/r/1081/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@303509 f38db490-d61c-443f-a65b-d21fe96a405b
If someone wishes to do so, it is trivial to set your own default when running
the configure script.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@303418 f38db490-d61c-443f-a65b-d21fe96a405b
Discussion on #asterisk on 2011-01-19:
(02:07:03 PM) boch: i wonder how to cycle all entries in a tree
(02:07:11 PM) leifmadsen: use While()
(02:07:17 PM) leifmadsen: you need to know the tree structure already though
(02:07:36 PM) boch: what you mean?
(02:09:02 PM) leifmadsen: you need to know the structure prior to looping, because you can't just return the structure from the dialplan
(02:09:43 PM) leifmadsen: the only way I can think of doing that is via something like writing the output of: asterisk -rx "database show" to a file, then looping through that to know the structure of the database and check everything
(02:09:59 PM) leifmadsen: but at that point you're better off just using either a relational database or an external script
(02:10:13 PM) boch: for example i need to know all entries in the tree
(02:10:15 PM) boch: got it
(02:10:20 PM) leifmadsen: exactly
(02:10:22 PM) leifmadsen: that's the problem
(02:10:22 PM) boch: thank you
(02:13:09 PM) mateu: yeah, i'm surprised there isn't something from the dialplan like 'database show family' so one can get all keys in a family to loop over.
(02:15:35 PM) leifmadsen: database shows everything
(02:16:22 PM) mateu: i mean something from the dial plan that mimics 'database show <family>'
(02:16:41 PM) leifmadsen: guess no one has found that important enough to program :)
(02:16:52 PM) leifmadsen: at that point you should probably just use a relational database...
(02:17:10 PM) mateu: i dunno
(02:17:16 PM) mateu: seems pretty basic to me.
(02:17:16 PM) leifmadsen: me either
(02:17:19 PM) leifmadsen: sure does
(02:17:24 PM) leifmadsen: no one has programmed it though
(02:17:28 PM) ***leifmadsen shrugs
(02:17:43 PM) mateu: ok, well at least we know how it currently stands. thanks leifmadsen
(02:28:52 PM) Corydon76-home: leifmadsen: something like HASHKEYS() ?
(02:30:11 PM) leifmadsen: Corydon76-home: ummm, I was thinking more like DUNDI_QUERY() and DUNDI_RESULT()
(02:30:31 PM) leifmadsen: although HASHKEYS() might work
(02:30:58 PM) leifmadsen: actually ya, looking at it, similar to HASHKEYS()
(02:31:01 PM) leifmadsen: DBKEYS() I guess?
(02:31:45 PM) Corydon76-home: So with no argument, retrieves families, with an argument, retrieves keys of that family?
(02:34:02 PM) leifmadsen: ya
(02:34:16 PM) leifmadsen: how would you iterate through layers of them?
(02:34:30 PM) leifmadsen: i.e. family/key/key/key ?
(02:34:43 PM) Corydon76-home: Essentially, yes
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r303153 | rmudgett | 2011-01-20 14:31:20 -0600 (Thu, 20 Jan 2011) | 22 lines
Merged revision 303098 from
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r303098 | rmudgett | 2011-01-20 12:11:45 -0600 (Thu, 20 Jan 2011) | 15 lines
CC_INTERFACES does not get built correctly with local channels.
If local channels are used with CCSS, CC_INTERFACES gets garbage prepended
to it so the CC recall fails. Also CC_INTERFACES gets "&(null)" appended
to it.
* Initialize the buffer to eliminate the prepended garbage.
* Filter out the empty interface strings to eliminate the latter.
* Added a diagnostic message if the CC_INTERFACES is ever empty.
JIRA ABE-2740
JIRA SWP-2848
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r303107 | sruffell | 2011-01-20 13:57:31 -0600 (Thu, 20 Jan 2011) | 23 lines
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r303106 | sruffell | 2011-01-20 13:56:34 -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 #18262)
Reported by: francesco_r
Patches:
patch_park_moh-trunk-2.txt uploaded by cjacobsen (license 1029)
Tested by: francesco_r, rfrantik, one47
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r303009 | jpeeler | 2011-01-20 11:10:32 -0600 (Thu, 20 Jan 2011) | 21 lines
Merged revisions 303008 via svnmerge from
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r303008 | jpeeler | 2011-01-20 11:07:44 -0600 (Thu, 20 Jan 2011) | 14 lines
Merged revisions 303007 via svnmerge from
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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
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r302918 | lmadsen | 2011-01-20 09:45:39 -0600 (Thu, 20 Jan 2011) | 16 lines
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r302917 | lmadsen | 2011-01-20 09:42:05 -0600 (Thu, 20 Jan 2011) | 8 lines
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)
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r302785 | russell | 2011-01-19 16:35:15 -0600 (Wed, 19 Jan 2011) | 15 lines
Resolve a memory leak with the manager interface is disabled.
The intent of this check as it stands in previous versions of Asterisk was to
check if there are any active sessions. If there were no sessions, then the
function would return immediately and not bother with queueing up the manager
event to be processed. Since the conversion of storing sessions in an astobj2
container, this check will always pass. I changed it to go back to checking
what was intended.
The side effect of this was that if the AMI is disabled, the manager event
queue is populated anyway, but the code that runs to clear out the queue
never runs. A producer with no consumer is a bad thing.
Reported internally by kmorgan.
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r302713 | rmudgett | 2011-01-19 15:29:22 -0600 (Wed, 19 Jan 2011) | 29 lines
Merged revisions 302693 via svnmerge from
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r302693 | rmudgett | 2011-01-19 15:25:41 -0600 (Wed, 19 Jan 2011) | 22 lines
Merged revisions 302671 via svnmerge from
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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
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r302634 | tilghman | 2011-01-19 14:24:57 -0600 (Wed, 19 Jan 2011) | 22 lines
Merged revisions 302599 via svnmerge from
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r302599 | tilghman | 2011-01-19 14:13:24 -0600 (Wed, 19 Jan 2011) | 15 lines
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
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r302555 | seanbright | 2011-01-19 14:03:32 -0500 (Wed, 19 Jan 2011) | 14 lines
Merged revisions 302554 via svnmerge from
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r302554 | seanbright | 2011-01-19 14:02:29 -0500 (Wed, 19 Jan 2011) | 7 lines
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)
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r302549 | seanbright | 2011-01-19 13:43:11 -0500 (Wed, 19 Jan 2011) | 17 lines
Merged revisions 302548 via svnmerge from
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r302548 | seanbright | 2011-01-19 13:37:09 -0500 (Wed, 19 Jan 2011) | 10 lines
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)
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r302412 | seanbright | 2011-01-19 10:31:39 -0500 (Wed, 19 Jan 2011) | 10 lines
Use appropriate type for requested format in chan_local.
We were passing and storing the requested format as an int instead of format_t
resulting in truncation.
(closes issue #18238)
Reported by: whizemen
Patches:
0018238_speex16.patch uploaded by whizemen (license 1143)
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r302266 | jpeeler | 2011-01-18 14:19:57 -0600 (Tue, 18 Jan 2011) | 34 lines
Merged revisions 302265 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r302265 | jpeeler | 2011-01-18 14:13:52 -0600 (Tue, 18 Jan 2011) | 27 lines
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/
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r302267 | russell | 2011-01-18 14:19:57 -0600 (Tue, 18 Jan 2011) | 5 lines
Don't enable AO2_DEBUG by default if AST_DEVMODE is on.
AO2_DEBUG is not important and is causing a false compiler warning to be
generated on my Ubuntu Natty dev box.
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r302174 | rmudgett | 2011-01-18 12:11:43 -0600 (Tue, 18 Jan 2011) | 102 lines
Merged revisions 302173 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r302173 | rmudgett | 2011-01-18 12:07:15 -0600 (Tue, 18 Jan 2011) | 95 lines
Merged revisions 302172 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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/
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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 #17403)
Reported by: one47
Patches:
sip_codecs_simplified4 uploaded by one47 (license 23)
Tested by: one47, falves11
Review: https://reviewboard.asterisk.org/r/967/
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r301946 | rmudgett | 2011-01-14 15:09:57 -0600 (Fri, 14 Jan 2011) | 13 lines
Deadlock between dahdi_request() and pri_dchannel() processing an incomming call.
The sig_pri_new_ast_channel() is called with the channel private lock held
when pri_dchannel() calls it and no channel private lock held when
dahdi_request() calls it. The use of pri_grab() in
sig_pri_new_ast_channel() could leave the channel private lock held when
it returns if the lock was not held before calling it.
Make sig_pri_new_ast_channel() just lock the PRI span lock instead of
using pri_grab(). It is safe to do this because dahdi_request() does not
have the channel private lock and the deadlock potential with the PRI span
lock is only between pri_dchannel() and other threads.
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r301790 | jpeeler | 2011-01-14 11:32:52 -0600 (Fri, 14 Jan 2011) | 42 lines
Resolve deadlock involving REFER.
Two fixes:
1) One must always have the private unlocked before calling
pbx_builtin_setvar_helper to not invalidate locking order since it locks the
channel.
2) Unlock the channel before calling pbx_find_extension, which starts and stops
autoservice during the lookup. The problem scenario as illustrated by the
reporter:
Thread: do_monitor
-----------------------
handle_request_do
handle_incoming
handle_request_refer
ast_parking_ext_valid
pbx_find_extension
ast_autoservice_stop
while (chan_list_state == as_chan_list_state) { usleep(1000); }
Thread: autoservice_run
-----------------------
autoservice_run
chan = ast_waitfor_n
ast_waitfor_nandfds
ast_waitfor_nandfds_classic / simple / complex (depending on your system)
ast_channel_lock(c[x]);
handle_request_do and schedule_process_request_queue locks the owner
if it exists. The autoservice thread is waiting for the channel lock, which
wasn't ever released since the do_monitor thread was waiting for autoservice
operations to complete. Solved by unlocking the channel but keeping a reference
to guarantee safety.
(closes issue #18403)
Reported by: jthurman
Patches:
20110103-blind_deadlock.diff uploaded by jthurman (license 614)
issue18403.patch uploaded by jpeeler (license 325)
Tested by: jthurman
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r301595 | mnicholson | 2011-01-12 12:51:37 -0600 (Wed, 12 Jan 2011) | 22 lines
Merged revisions 301594 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r301594 | mnicholson | 2011-01-12 12:50:31 -0600 (Wed, 12 Jan 2011) | 15 lines
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
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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
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r301504 | jpeeler | 2011-01-12 12:12:08 -0600 (Wed, 12 Jan 2011) | 26 lines
Merged revisions 301503 via svnmerge from
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r301503 | jpeeler | 2011-01-12 12:11:49 -0600 (Wed, 12 Jan 2011) | 19 lines
Merged revisions 301502 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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the warning, if no user interface for menuselect warning was found is not right.
you have to rerun configure before make menuselect after installing a proper user interface.
(closes issue 0018594)
Reported by: Dovid
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@301445 f38db490-d61c-443f-a65b-d21fe96a405b
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r301221 | pabelanger | 2011-01-09 16:40:34 -0500 (Sun, 09 Jan 2011) | 21 lines
Merged revisions 301220 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r301220 | pabelanger | 2011-01-09 16:38:24 -0500 (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 #18589)
Reported by: pabelanger
Patches:
issue18589.patch uploaded by pabelanger (license 224)
Tested by: pabelanger
Review: https://reviewboard.asterisk.org/r/1074/
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r301134 | rmudgett | 2011-01-07 19:11:31 -0600 (Fri, 07 Jan 2011) | 7 lines
The DTMF attended transfer feature cannot callback a chan_dahdi BRI phone.
The DAHDI ISDN channel name is not dialable.
Make a channel name like DAHDI/i3/400-12 dialable when the sequence number
is stripped off of the name.
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r301047 | jpeeler | 2011-01-07 13:58:30 -0600 (Fri, 07 Jan 2011) | 15 lines
Merged revisions 301046 via svnmerge from
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r301046 | jpeeler | 2011-01-07 13:57:42 -0600 (Fri, 07 Jan 2011) | 8 lines
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
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r300955 | jpeeler | 2011-01-07 11:24:14 -0600 (Fri, 07 Jan 2011) | 21 lines
Merged revisions 300951 via svnmerge from
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r300951 | jpeeler | 2011-01-07 11:23:37 -0600 (Fri, 07 Jan 2011) | 14 lines
Merged revisions 300918 via svnmerge from
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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
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If your postgres connection died suddenly in between res_config_pgsql
queries, the next query will fail because the query is executed on a
disconnected/disconnecting handle. The query is abandoned and is
returned from in error.
Now we will reconnect and try again if a query was run on a
disconnected connection.
(closes issue #18071)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@300882 f38db490-d61c-443f-a65b-d21fe96a405b
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r300714 | rmudgett | 2011-01-05 14:54:21 -0600 (Wed, 05 Jan 2011) | 21 lines
Merged revision 300711 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
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r300711 | rmudgett | 2011-01-05 13:43:55 -0600 (Wed, 05 Jan 2011) | 14 lines
A call retrieved from hold may wind up with no audio.
If the retrieved call is natively bridged then the call may not have any
audio path. The following warning message is given:
"Failed to add <dfd> to conference <chan>/<chan>: Invalid argument".
* Open the media on a B channel when pri_fixup_principle() moves the call
from a no_b_channel channel to a real channel.
* Added lock protection while pri_fixup_principle() moves a call from one
private structure to another.
* Made some pri_fixup_principle() messages more meaningful.
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r300301 | twilson | 2011-01-04 11:54:41 -0600 (Tue, 04 Jan 2011) | 29 lines
Merged revisions 300298 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r300298 | twilson | 2011-01-04 11:37:26 -0600 (Tue, 04 Jan 2011) | 22 lines
Merged revisions 300216 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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/
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Added the moh_signaling option to specify what to do when the channel's
bridged peer puts the ISDN channel on and off of hold.
Implemented as a FSM to control libpri ISDN signaling when the bridged
peer places the channel on and off of hold with the AST_CONTROL_HOLD and
AST_CONTROL_UNHOLD control frames.
JIRA SWP-2687
JIRA ABE-2691
Review: https://reviewboard.asterisk.org/r/1063/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@300212 f38db490-d61c-443f-a65b-d21fe96a405b
playing_silence was not initialized with the struct
was initialized, it was being set after the fact
which caused problems if something that relied on
playing_silence being set was called too quickly
(closes issue #18430)
Reported by: stevebrandli
Patches:
externalivr.patch uploaded by thedavidfactor (license 903)
Tested by: thedavidfactor, stevebrandli
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@300121 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r300082 | lmadsen | 2011-01-03 07:14:25 -0600 (Mon, 03 Jan 2011) | 11 lines
Increase side of mapping response field.
I've increased the size of the response field in a DUNDi mapping because of
some documentation I'm writing. Previously it was set to AST_MAX_EXTENSION which
is only 80 characters, which is far too small when you're using some dialplan
functions to craft a response. The example I'm using is:
extensions =>
RegisteredDevices,0,SIP,dundi:very_awesome_password/${IF($[${DB_EXISTS(phones/${NUMBER}/device)}]?${DB(phones/${NUMBER}/device)}:None)},nopartial
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Change order of sending Terminal Capability Set and MasterSlave
Determination packets, MSD send when TCS exchange procedure is done
(we send tcs ack to remote and we have remote tcs ack already
or we receive tcs ack from remote and we have send our tcs ack to
remote already). Some endpoints can work in this sequence only,
i suggest they can't work with both (tcs and msd) exchange procedures
simultaneously.
Also changed StartH245 facility message sending. It send on
incoming calls only due to some endpoints can't proccess properly
this facility messages on their incoming calls.
(closes issue #18433)
Reported by: MrHanMan
Patches:
tcs-msd-h245-3.patch uploaded by may213 (license 454)
Tested by: MrHanMan, may213
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@299715 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r299449 | tilghman | 2010-12-22 14:05:02 -0600 (Wed, 22 Dec 2010) | 15 lines
Merged revisions 299448 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r299448 | tilghman | 2010-12-22 14:03:30 -0600 (Wed, 22 Dec 2010) | 8 lines
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
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r299405 | rmudgett | 2010-12-21 20:10:39 -0600 (Tue, 21 Dec 2010) | 17 lines
Chan_dahdi sends an empty COLP on the bridged channel.
Chan_dahdi always inserts a connected party IE when you call from one
dahdi channel to another dahdi channel, even if no such information was
received on the 2nd channel. This clears the display of many phones.
* Removed leftover artifact from before the valid flag was added.
* Updated all of the channel's caller id information with the new
connected line information instead of just the string parts.
(closes issue #18508)
Reported by: wimpy
Patches:
issue18508_trunk.patch uploaded by rmudgett (license 664)
Tested by: wimpy, rmudgett
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r299353 | mnicholson | 2010-12-21 09:25:03 -0600 (Tue, 21 Dec 2010) | 30 lines
Merged revisions 299242 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r299242 | mnicholson | 2010-12-20 15:25:35 -0600 (Mon, 20 Dec 2010) | 23 lines
Merged revisions 299194,299198,299220 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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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.
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r299220 | mnicholson | 2010-12-20 15:21:39 -0600 (Mon, 20 Dec 2010) | 4 lines
Use ast_free() instead of free()
ABE-2656
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r299248 | mmichelson | 2010-12-20 15:38:30 -0600 (Mon, 20 Dec 2010) | 20 lines
Fix a couple of CCSS issues.
* Make sure to allocate a cc_params structure
when creating autopeers.
* Use sip_uri_cmp when retrieving SIP CC agents
and monitors in case parameters appear in the
URI.
(closes issue #18504)
Reported by: kkm
(closes issue #18338)
Reported by: GeorgeKonopacki
Patches:
18338.diff uploaded by mmichelson (license 60)
Tested by: GeorgeKonopacki
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r299131 | tilghman | 2010-12-20 11:47:10 -0600 (Mon, 20 Dec 2010) | 18 lines
Merged revisions 299130 via svnmerge from
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r299130 | tilghman | 2010-12-20 11:41:24 -0600 (Mon, 20 Dec 2010) | 11 lines
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
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Previously, I had added the ast_sched_thread stuff that was a generic scheduler
thread implementation. However, if you used it, it required using different
functions for modifying scheduler contents. This patch reworks how this is
done and just allows you to optionally start a thread on the original scheduler
context structure that has always been there. This makes it trivial to switch
to the generic scheduler thread implementation without having to touch any of
the other code that adds or removes scheduler entries.
In passing, I made some naming tweaks to add ast_ prefixes where they were not
there before.
Review: https://reviewboard.asterisk.org/r/1007/
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r298960 | tilghman | 2010-12-17 17:52:04 -0600 (Fri, 17 Dec 2010) | 20 lines
Merged revisions 298957 via svnmerge from
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r298957 | tilghman | 2010-12-17 17:30:55 -0600 (Fri, 17 Dec 2010) | 13 lines
Merged revisions 298905 via svnmerge from
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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.
Review: https://reviewboard.asterisk.org/r/1055/
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r298598 | jpeeler | 2010-12-16 14:51:44 -0600 (Thu, 16 Dec 2010) | 21 lines
Merged revisions 298597 via svnmerge from
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r298597 | jpeeler | 2010-12-16 14:49:33 -0600 (Thu, 16 Dec 2010) | 14 lines
Merged revisions 298596 via svnmerge from
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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
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r298539 | tilghman | 2010-12-16 03:28:17 -0600 (Thu, 16 Dec 2010) | 8 lines
Ensure the ipaddr field in realtime is large enough to handle IPv6 addresses.
(closes issue #18464)
Reported by: IgorG
Patches:
realtime_ipv6store.diff uploaded by IgorG (license 20)
(plus a few additional lines by tilghman)
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r298195 | rmudgett | 2010-12-13 11:11:43 -0600 (Mon, 13 Dec 2010) | 33 lines
Merged revisions 298194 via svnmerge from
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r298194 | rmudgett | 2010-12-13 11:04:41 -0600 (Mon, 13 Dec 2010) | 26 lines
Merged revisions 298193 via svnmerge from
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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
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Already had the pthread ID which is not the same. The most obvious enhancement
is in the "core show threads" output. As stated in the utils header, if the
platform isn't supported -1 is reported (instead of the process ID previously).
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@298137 f38db490-d61c-443f-a65b-d21fe96a405b
Don't use GK ID if it's not presented in GK replies
Extract GK ID not only in GK confirm but in GK register confirm also
(closes issue #18401)
Reported by: MrHanMan
Patches:
no-gkid-2.patch uploaded by may213 (license 454)
Tested by: may213, MrHanMan
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@298100 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r297965 | twilson | 2010-12-09 16:18:19 -0600 (Thu, 09 Dec 2010) | 28 lines
Merged revisions 297960 via svnmerge from
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r297960 | twilson | 2010-12-09 16:10:31 -0600 (Thu, 09 Dec 2010) | 21 lines
Merged revisions 297959 via svnmerge from
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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/
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r297957 | dvossel | 2010-12-09 15:32:20 -0600 (Thu, 09 Dec 2010) | 11 lines
Fixes issue with outbound google voice calls not working.
Thanks to az1234 and nevermind_quack for their input in helping debug the issue.
(closes issue #18412)
Reported by: nevermind_quack
Patches:
fix uploaded by dvossel (license 671)
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r297825 | jpeeler | 2010-12-07 16:59:30 -0600 (Tue, 07 Dec 2010) | 26 lines
Merged revisions 297824 via svnmerge from
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r297824 | jpeeler | 2010-12-07 16:58:54 -0600 (Tue, 07 Dec 2010) | 19 lines
Merged revisions 297823 via svnmerge from
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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
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r297607 | jpeeler | 2010-12-06 16:06:37 -0600 (Mon, 06 Dec 2010) | 25 lines
Merged revisions 297605 via svnmerge from
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r297605 | jpeeler | 2010-12-06 16:03:04 -0600 (Mon, 06 Dec 2010) | 18 lines
Merged revisions 297603 via svnmerge from
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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
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r297157 | mnicholson | 2010-12-01 13:47:33 -0600 (Wed, 01 Dec 2010) | 2 lines
Changed some NOTICE and WARNING messages to DEBUG messages.
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r297486 | mnicholson | 2010-12-02 15:30:47 -0600 (Thu, 02 Dec 2010) | 6 lines
Add support for reserving a fax session before answering the channel.
Note: this change breaks ABI compatibility.
FAX-217
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r297495 | mnicholson | 2010-12-03 09:21:52 -0600 (Fri, 03 Dec 2010) | 4 lines
Print a DEBUG message instead of a WARNING message when the selected fax tech does not support reserving sessions.
Answer the channel before quering it for t.38 support. This is necessary for the query to work properly over local channels.
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r297312 | twilson | 2010-12-02 12:13:49 -0600 (Thu, 02 Dec 2010) | 28 lines
Merged revisions 297311 via svnmerge from
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r297311 | twilson | 2010-12-02 12:07:39 -0600 (Thu, 02 Dec 2010) | 21 lines
Merged revisions 297310 via svnmerge from
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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/
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r297245 | russell | 2010-12-02 07:20:19 -0600 (Thu, 02 Dec 2010) | 20 lines
Merged revisions 297229 via svnmerge from
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r297229 | russell | 2010-12-02 07:16:47 -0600 (Thu, 02 Dec 2010) | 13 lines
Merged revisions 297228 via svnmerge from
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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.
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r297075 | jpeeler | 2010-12-01 11:53:13 -0600 (Wed, 01 Dec 2010) | 37 lines
Merged revisions 297073 via svnmerge from
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r297073 | jpeeler | 2010-12-01 11:52:46 -0600 (Wed, 01 Dec 2010) | 30 lines
Merged revisions 297072 via svnmerge from
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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
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by splitting up devices from hints into an own ao2_container the callback to
get these devices for statechange handling is faster.
with this changes the length of a device used in a hint isnt longer restricted
to 80 characters.
Tests showed that calling handle_statechange is 40 times faster if no hints
are used and 25 times faster if there are any hints.
(closes issue #17928)
Reported by: mdu113
Tested by: schmidts
Review: https://reviewboard.asterisk.org/r/1003/
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r296628 | russell | 2010-11-29 15:26:44 -0600 (Mon, 29 Nov 2010) | 6 lines
Complete some error handling in transmit_publish() in chan_sip.c.
This error handling block caught my eye. It was missing a couple of things,
but it should be safe now. Thanks to mmichelson for the quick peer review
on IRC.
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r296582 | rmudgett | 2010-11-29 14:46:03 -0600 (Mon, 29 Nov 2010) | 24 lines
Merged revision 296575 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
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r296575 | rmudgett | 2010-11-29 14:27:37 -0600 (Mon, 29 Nov 2010) | 13 lines
Invalid mISDN PTMP redirecting signaling as TE towards NT.
The mISDN PTMP redirection signaling (NOTIFY redirecting number and
notification code, SETUP redirecting number) is also sent in PTMP/TE mode.
It should only apply in PTMP/NT mode. The call setup proceeds but the
network (Deutsche Telekom) reacts with ugly ISDN STATUS messages.
Also don't send the redirecting number ie when PTP is also sending the
DivertingLegInformation2 facility. The redirecting number ie is redundant
and the network (Deutsche Telekom) complains about it.
Patches:
abe_2651_v4.patch uploaded by rmudgett (license 664)
JIRA ABE-2651
JIRA SWP-2537
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r296534 | tilghman | 2010-11-29 01:28:44 -0600 (Mon, 29 Nov 2010) | 20 lines
Merged revisions 296533 via svnmerge from
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r296533 | tilghman | 2010-11-29 01:27:09 -0600 (Mon, 29 Nov 2010) | 13 lines
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
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r296352 | marquis | 2010-11-26 13:19:02 -0500 (Fri, 26 Nov 2010) | 12 lines
Fix reloading of peer when a user is requested.
Prevent peer reloading from causing multiple MWI subscriptions to be created when using realtime. This had the effect of sending one NOTIFY for every time a sip peer made a call, in one case eventually overwhelming the phone and causing it to reboot.
(closes issue #18342)
Reported by: nivek
Patches:
issue0018342p1.patch uploaded by nivek (license 636)
Tested by: nivek
Review: https://reviewboard.asterisk.org/r/1029/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@296353 f38db490-d61c-443f-a65b-d21fe96a405b
Added option v(mailbox@[context]) which tells MeetMe where to look for a users greet file. If one does not exist it clears the v option and defers to the functionality of i/I as/if set by the MeetMe() command.
Review: https://reviewboard.asterisk.org/r/1009/
(closes issue #18297)
Reported by: parisioa
Patches:
meetme_final_patch_v.diff uploaded by parisioa (license 1153)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@296249 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r296230 | russell | 2010-11-24 17:29:44 -0600 (Wed, 24 Nov 2010) | 20 lines
Merged revisions 296221 via svnmerge from
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r296221 | russell | 2010-11-24 17:28:19 -0600 (Wed, 24 Nov 2010) | 13 lines
Merged revisions 296213 via svnmerge from
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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.
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r296167 | rmudgett | 2010-11-24 16:49:48 -0600 (Wed, 24 Nov 2010) | 57 lines
Merged revisions 296166 via svnmerge from
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r296166 | rmudgett | 2010-11-24 16:42:45 -0600 (Wed, 24 Nov 2010) | 50 lines
Merged revisions 296165 via svnmerge from
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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.
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r296084 | russell | 2010-11-24 14:23:46 -0600 (Wed, 24 Nov 2010) | 26 lines
Merged revisions 296083 via svnmerge from
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r296083 | russell | 2010-11-24 14:23:11 -0600 (Wed, 24 Nov 2010) | 19 lines
Merged revisions 296082 via svnmerge from
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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. :-)
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r296002 | russell | 2010-11-24 11:13:08 -0600 (Wed, 24 Nov 2010) | 52 lines
Merged revisions 296001 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r296001 | russell | 2010-11-24 11:03:16 -0600 (Wed, 24 Nov 2010) | 45 lines
Merged revisions 296000 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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r295866 | rmudgett | 2010-11-22 13:36:10 -0600 (Mon, 22 Nov 2010) | 60 lines
Merged revisions 295843 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r295843 | rmudgett | 2010-11-22 13:28:23 -0600 (Mon, 22 Nov 2010) | 53 lines
Merged revisions 295790 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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/
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r295747 | rmudgett | 2010-11-19 21:11:15 -0600 (Fri, 19 Nov 2010) | 13 lines
One way audio before answering call waiting call on analog port.
* Analog call waiting Caller ID spills could get stuck resulting in one
way audio until the waiting call is answered. This only happens on the
second (and later) call waiting call if the active call is not the first
call.
* The CLI/AMI "dahdi show channel" command could report the wrong channel
information.
Must keep the struct analog_pvt.owner and struct dahdi_pvt.owner pointer
in sync.
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r295711 | russell | 2010-11-19 18:50:00 -0600 (Fri, 19 Nov 2010) | 36 lines
Merged revisions 295710 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r295710 | russell | 2010-11-19 18:45:51 -0600 (Fri, 19 Nov 2010) | 29 lines
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/
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Add the ability to play a sound file, listen for more than just one digit,
specify
escape characters. Backwards compatible (to work with only timeout specified).
(closes issue #15531)
Reported by: diLLec
Patches:
asterisk-res_agi-203638-patched.patch uploaded by diLLec (license 839)
Tested by: diLLec, espiceland
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@295554 f38db490-d61c-443f-a65b-d21fe96a405b
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r295516 | rmudgett | 2010-11-19 10:47:11 -0600 (Fri, 19 Nov 2010) | 13 lines
Bring sig_analog extraction more into alignment with orig-trunk/v1.6.2 chan_dahdi.
* Restore SMDI support.
* Fixed initial value of struct analog_pvt.use_callerid. It may get
forced on depending upon other config options.
* Call analog_dnd() instead of manual inlined code.
* Removed unused struct analog_pvt.usedistinctiveringdetection.
* Removed the struct analog_pvt.unknown_alarm flag. It was really the
struct analog_pvt.inalarm flag.
* Use ast_debug() instead of ast_log(LOG_DEBUG).
* Rename several function's index variable to idx.
* Some formatting tweaks.
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r294905 | jpeeler | 2010-11-12 14:52:06 -0600 (Fri, 12 Nov 2010) | 30 lines
Merged revisions 294904 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r294904 | jpeeler | 2010-11-12 14:51:15 -0600 (Fri, 12 Nov 2010) | 23 lines
Merged revisions 294903 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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r294823 | rmudgett | 2010-11-11 20:45:22 -0600 (Thu, 11 Nov 2010) | 25 lines
Merged revisions 294822 via svnmerge from
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r294822 | rmudgett | 2010-11-11 20:44:12 -0600 (Thu, 11 Nov 2010) | 18 lines
Merged revisions 294821 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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r294740 | russell | 2010-11-11 16:13:38 -0600 (Thu, 11 Nov 2010) | 11 lines
Remove most of the contents of the doc dir in favor of the wiki content.
This merge does the following things:
* Removes most of the contents from the doc/ directory in favor
of the wiki - http://wiki.asterisk.org/
* Updates the build_tools/prep_tarball script to know how to export
the contents of the wiki in both PDF and plain text formats so that
the documentation is still included in Asterisk release tarballs.
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r294734 | jpeeler | 2010-11-11 15:58:25 -0600 (Thu, 11 Nov 2010) | 32 lines
Merged revisions 294733 via svnmerge from
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r294733 | jpeeler | 2010-11-11 15:57:22 -0600 (Thu, 11 Nov 2010) | 25 lines
Merged revisions 294688 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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r294535 | russell | 2010-11-10 08:14:51 -0600 (Wed, 10 Nov 2010) | 5 lines
Tweak a couple of CLI commands back to their original form.
The "module" in this case is two parts, so there are two words before
the verb of the CLI command.
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r294349 | rmudgett | 2010-11-09 10:55:32 -0600 (Tue, 09 Nov 2010) | 17 lines
Analog lines do not transfer CONNECTED LINE or execute the interception macros.
Add connected line update for sig_analog transfers and simplify the
corresponding sig_pri and chan_misdn transfer code.
Note that if you create a three-way call in sig_analog before transferring
the call, the distinction of the caller/callee interception macros make
little sense. The interception macro writer needs to be prepared for
either caller/callee macro to be executed. The current implementation
swaps which caller/callee interception macro is executed after a three-way
call is created.
Review: https://reviewboard.asterisk.org/r/996/
JIRA ABE-2589
JIRA SWP-2372
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r294278 | jpeeler | 2010-11-08 15:59:45 -0600 (Mon, 08 Nov 2010) | 23 lines
Merged revisions 294277 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r294277 | jpeeler | 2010-11-08 15:58:13 -0600 (Mon, 08 Nov 2010) | 16 lines
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!
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r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines
valgrind reported references to freed memory during a mISDN hangup collision.
Bad things have been happening in chan_misdn because the chan_misdn
channel private struct chan_list is not protected from reentrancy. Hangup
collisions have be causing read and write accesses to freed memory.
Converted chan_misdn struct chan_list to an ao2 object for its reference
counting feature.
**********
Removed an impediment to converting chan_list to an ao2 object.
The use of the other_ch member in chan_list is shaky at best. It is set
if the incoming and outgoing call legs are mISDN. The use of the other_ch
member goes against the Asterisk architecture and can even cause problems.
1) It is used to disable echo cancellation. This could be bad if the call
is forked and the winning call leg is not mISDN or the winning call leg is
not the last mISDN channel called by the fork. The other_ch would become
a dangling pointer.
2) It is used when the far end is alerting to hear the far end's inband
audio instead of Asterisk's generated ringback tone. This is bad if the
call is forked. You would only hear the last forked mISDN channel and it
may not be ringing yet.
The other_ch would become a dangling pointer if the call is later
transferred.
**********
JIRA SWP-2423
JIRA ABE-2614
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RFC3261 section 12 about dialog creation says an INVITE transaction
results in an established dialog once it receives the 200 OK response.
It is possible to receive multiple differing 200 OK responses for a
single outbound INVITE Request, and this should result in establishing
multiple dialogs.
This patch allows for all differing 200 OK responses to an INVITE request
to establish a separate dialog, but only the first dialog is kept. All other
resulting dialogs from the initial request are immediately ACKed and then
immediately terminated with a BYE request.
Review: https://reviewboard.asterisk.org/r/946/
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r293970 | sruffell | 2010-11-04 19:07:11 -0500 (Thu, 04 Nov 2010) | 32 lines
Merged revisions 293969 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r293969 | sruffell | 2010-11-04 19:06:02 -0500 (Thu, 04 Nov 2010) | 25 lines
Merged revisions 293968 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
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r293803 | twilson | 2010-11-03 11:05:14 -0700 (Wed, 03 Nov 2010) | 25 lines
Avoid valgrind warnings for ast_rtp_instance_get_xxx_address
The documentation for ast_rtp_instance_get_(local/remote)_address stated that
they returned 0 for success and -1 on failure. Instead, they returned 0 if the
address structure passed in was already equivalent to the address instance
local/remote address or 1 otherwise. 90% of the calls to these functions
completely ignored the return address and passed in an uninitialized struct,
which would make valgrind complain even though the operation was technically
safe.
This patch fixes the documentation and converts the get_xxx_address functions
to void since all they really do is copy the address and cannot fail.
Additionally two new functions
(ast_rtp_instance_get_and_cmp_(local/remote)_address) are created for the 3
times where the return value was actually checked. The
get_and_cmp_local_address function is currently unused, but exists for the sake
of symmetry.
The only functional change as a result of this change is that we will not do an
ast_sockaddr_cmp() on (mostly uninitialized) addresses before doing the
ast_sockaddr_copy() in the get_*_address functions. So, even though it is an
API change, it shouldn't have a noticeable change in behavior.
Review: https://reviewboard.asterisk.org/r/995/
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r293807 | rmudgett | 2010-11-03 13:35:19 -0500 (Wed, 03 Nov 2010) | 34 lines
Merged revisions 293806 via svnmerge from
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r293806 | rmudgett | 2010-11-03 13:31:57 -0500 (Wed, 03 Nov 2010) | 27 lines
Merged revisions 293805 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
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r293648 | rmudgett | 2010-11-02 16:29:25 -0500 (Tue, 02 Nov 2010) | 20 lines
Merged revisions 293647 via svnmerge from
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r293647 | rmudgett | 2010-11-02 16:26:30 -0500 (Tue, 02 Nov 2010) | 13 lines
Merged revisions 293639 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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>)".
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r293530 | rmudgett | 2010-11-01 12:29:30 -0500 (Mon, 01 Nov 2010) | 10 lines
Analog 3-way call would not connect all parties if one was using sig_pri.
Also the "dahdi show channel" would not show the correct 3-way call
status.
* Synchronized the inthreeway flag between chan_dahdi and sig_analog.
* Fixed a my_set_linear_mode() sign error and made take an analog sub
channel enum.
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r293305 | jpeeler | 2010-10-29 16:48:38 -0500 (Fri, 29 Oct 2010) | 9 lines
Modify sip_setoption to not complain about unknown options.
This now behaves just like the other setoption callbacks. For the curious the
offending option for the reporter was AST_OPTION_CHANNEL_WRITE which was getting
passed due to a fix for chan_local in 286189.
(closes issue #17985)
Reported by: globalnetinc
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r293119 | jpeeler | 2010-10-26 13:49:08 -0500 (Tue, 26 Oct 2010) | 43 lines
Merged revisions 293118 via svnmerge from
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r293118 | jpeeler | 2010-10-26 13:33:24 -0500 (Tue, 26 Oct 2010) | 36 lines
Merged revisions 293004 via svnmerge from
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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
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r292868 | dvossel | 2010-10-25 14:07:50 -0500 (Mon, 25 Oct 2010) | 39 lines
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r292867 | dvossel | 2010-10-25 14:06:21 -0500 (Mon, 25 Oct 2010) | 32 lines
Merged revisions 292866 via svnmerge from
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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.
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r292787 | lmadsen | 2010-10-22 16:28:43 -0500 (Fri, 22 Oct 2010) | 21 lines
Merged revisions 292786 via svnmerge from
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r292786 | lmadsen | 2010-10-22 16:16:12 -0500 (Fri, 22 Oct 2010) | 13 lines
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
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r292740 | twilson | 2010-10-22 09:49:34 -0700 (Fri, 22 Oct 2010) | 45 lines
Add TLS cert helper script
This script is useful for quickly generating self-signed CA, server, and client
certificates for use with Asterisk. It is still recommended to obtain
certificates from a recognized Certificate Authority and to develop an
understanding how SSL certificates work. Real security is hard work.
OPTIONS:
-h Show this message
-m Type of cert "client" or "server". Defaults to server.
-f Config filename (openssl config file format)
-c CA cert filename (creates new CA cert/key as ca.crt/ca.key if not passed)
-k CA key filename
-C Common name (cert field)
For a server cert, this should be the same address that clients
attempt to connect to. Usually this will be the Fully Qualified
Domain Name, but might be the IP of the server. For a CA or client
cert, it is merely informational. Make sure your certs have unique
common names.
-O Org name (cert field)
An informational string (company name)
-o Output filename base (defaults to asterisk)
-d Output directory (defaults to the current directory)
Example:
To create a CA and a server (pbx.mycompany.com) cert with output in /tmp:
ast_tls_cert -C pbx.mycompany.com -O "My Company" -d /tmp
This will create a CA cert and key as well as asterisk.pem and the the two
files that it is made from: asterisk.crt and asterisk.key. Copy asterisk.pem
and ca.crt somewhere (like /etc/asterisk) and set tlscertfile=/etc/asterisk.pem
and tlscafile=/etc/ca.crt. Since this is a self-signed key, many devices will
require you to import the ca.crt file as a trusted cert.
To create a client cert using the CA cert created by the example above:
ast_tls_cert -m client -c /tmp/ca.crt -k /tmp/ca.key -C "Joe User" -O \
"My Company" -d /tmp -o joe_user
This will create client.crt/key/pem in /tmp. Use this if your device supports
a client certificate. Make sure that you have the ca.crt file set up as
a tlscafile in the necessary Asterisk configs. Make backups of all .key files
in case you need them later.
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r292741 | mmichelson | 2010-10-22 12:09:52 -0500 (Fri, 22 Oct 2010) | 12 lines
Prevent multiple runs of event_sub_test from producing false failure results.
The array of test subscriptions was declared "static," meaning that the
data.count field would retain its value between runs of the test. After the
first test run, this would result in false reports of test failures.
I chose to just remove the "static" keyword from the structure since it's not
a huge deal to construct this structure during each run of the test. Another
alternative would have been to zero out the data.count fields of each test
subscription instead.
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r292704 | rmudgett | 2010-10-22 10:47:08 -0500 (Fri, 22 Oct 2010) | 19 lines
Connected line is not updated when chan_dahdi/sig_pri or chan_misdn transfers a call.
When a call is transfered by ECT or implicitly by disconnect in sig_pri or
implicitly by disconnect in chan_misdn, the connected line information is
not exchanged. The connected line interception macros also need to be
executed if defined.
The CALLER interception macro is executed for the held call.
The CALLEE interception macro is executed for the active/ringing call.
JIRA ABE-2589
JIRA SWP-2296
Patches:
abe_2589_c3bier.patch uploaded by rmudgett (license 664)
abe_2589_v1.8_v2.patch uploaded by rmudgett (license 664)
Review: https://reviewboard.asterisk.org/r/958/
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r292595 | dvossel | 2010-10-21 11:14:33 -0500 (Thu, 21 Oct 2010) | 14 lines
Fixes recursive lock problem in manager.c
It is possible for a AMI session to freeze because of invalid
use of recursive locks during the EVENT processing. This
patch removes the unnecessary locks.
(closes issue #18167)
Reported by: sustav
Patches:
manager_locking_v1.diff uploaded by dvossel (license 671)
Tested by: sustav
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r292489 | rmudgett | 2010-10-20 20:02:50 -0500 (Wed, 20 Oct 2010) | 7 lines
Send CONNECT_ACKNOWLEDGE for CIS calls too.
The originator of the Q.SIG call completion signaling link was not changed
to the active state when the CONNECT message came in. The T309 processing
would immediately kill the signaling link because it was not in the active
state.
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r292376 | tilghman | 2010-10-19 19:40:29 -0500 (Tue, 19 Oct 2010) | 5 lines
Oops. This module uses the generic timer and no longer uses DAHDI.
This causes a problem with the Solaris and other system builds that have gcc
4.1 (where optional_api is non-optional).
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r292309 | twilson | 2010-10-19 12:27:32 -0700 (Tue, 19 Oct 2010) | 10 lines
Add sip show peer info about crypto and remove dated comment
This patch adds information about the encryption setting to 'sip show
peers' and removes an out-of-date comment from res_srtp.c and instead
directs users to the proper documentation.
(closes issue #18140)
Reported by: chodorenko
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r292227 | jpeeler | 2010-10-18 16:55:46 -0500 (Mon, 18 Oct 2010) | 25 lines
Merged revisions 292226 via svnmerge from
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r292226 | jpeeler | 2010-10-18 16:54:38 -0500 (Mon, 18 Oct 2010) | 18 lines
Merged revisions 292223 via svnmerge from
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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
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r292188 | russell | 2010-10-18 14:50:04 -0500 (Mon, 18 Oct 2010) | 9 lines
Resolve some compiler errors in ast_sockaddr_is_any().
These errors came up once this function was used from within netsock2.c.
The errors were like the following:
netsock2.c:393: error: dereferencing pointer ‘({anonymous})’ does break strict-aliasing rules
The usage of a union here avoids this problem.
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r292050 | tzafrir | 2010-10-16 12:47:00 +0200 (ש', 16 אוק 2010) | 22 lines
Merged revisions 292049 via svnmerge from
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r292049 | tzafrir | 2010-10-16 12:03:04 +0200 (ש', 16 אוק 2010) | 15 lines
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/
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r292016 | twilson | 2010-10-15 16:40:56 -0500 (Fri, 15 Oct 2010) | 5 lines
Ref/unref res_srtp when we create/destroy a session
This avoids unhappy crashing when we try to 'core stop gracefully' and res_srtp
tries to unload before chan_sip does. Thanks, Russell!
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r291905 | twilson | 2010-10-15 09:39:58 -0700 (Fri, 15 Oct 2010) | 14 lines
Merged revisions 291904 via svnmerge from
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r291904 | twilson | 2010-10-15 09:16:57 -0700 (Fri, 15 Oct 2010) | 7 lines
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.
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r291827 | dvossel | 2010-10-14 16:27:42 -0500 (Thu, 14 Oct 2010) | 18 lines
Safer xml parsing, treat all clients the same, and better local candidate selection.
The gtalk channel driver was doing several unsafe operations
in regards to how it parsed incoming XML messages. I have cleaned
that code up so it should be much safer now.
We now treat all clients types the same. We have no reason to
distinguish between GMAIL and GOOGLE VOICE clients anymore because
they all work the same way.
I also modified how the local ip is found. If no bindaddress is provided
in the config file, we attempt to determine the local ip we
would use to connect to google.com. If that fails, then
we fall back to the ast_find_ourip() function as a last resort.
Using the new method makes it much less likely that we would ever
advertise a local RTP candidate as a loopback address.
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r291758 | pabelanger | 2010-10-14 11:15:12 -0400 (Thu, 14 Oct 2010) | 11 lines
Add the ability for ast_find_ourip to return IPv4, IPv6 or both.
While testing chan_gtalk I noticed jabber was using my IPv6 address
and not IPv4. When using bindaddr=0.0.0.0 it is possible for ast_find_ourip()
to return both IPv6 and IPv4 results. Adding a family parameter gives you
the ablility to choose.
Since jabber/gtalk/h323 do not support IPv6, we should only return IPv4 results.
Review: https://reviewboard.asterisk.org/r/973/
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r291656 | rmudgett | 2010-10-13 18:45:11 -0500 (Wed, 13 Oct 2010) | 34 lines
Merged revisions 291655 via svnmerge from
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r291655 | rmudgett | 2010-10-13 18:36:50 -0500 (Wed, 13 Oct 2010) | 27 lines
Merged revisions 291643 via svnmerge from
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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/
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r291581 | twilson | 2010-10-13 16:01:56 -0700 (Wed, 13 Oct 2010) | 35 lines
Merged revisions 291580 via svnmerge from
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r291580 | twilson | 2010-10-13 15:58:43 -0700 (Wed, 13 Oct 2010) | 28 lines
Merged revisions 291577 via svnmerge from
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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/
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r291575 | twilson | 2010-10-13 15:24:44 -0700 (Wed, 13 Oct 2010) | 8 lines
Add a simple AMI client web page
This patch uses the XML docs to parse all of the available AMI commands
and allows you to enter the command name and be presented with a form with
the available fields. You can then rapidly tab through the fields and submit
the command and view the response. It is much faster/easier than having to
use telnet for testing purposes.
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r291541 | rmudgett | 2010-10-13 15:21:02 -0500 (Wed, 13 Oct 2010) | 26 lines
The chan_dahdi faxdetect option only works for the first FAX call.
The chan_dahdi faxdetect option only works for the first call. After that
the option no longer works. The struct dahdi_pvt.callprogress member is
the encoded user config setting for the callprogress and faxdetect config
options. Changing this value alters the configuration for all following
calls until the chan_dahdi.conf file is reloaded.
* Fixed the chan_dahdi ast_channel_setoption callback to not change the
users faxdetect config setting except for the current call.
* Fixed the chan_dahdi ast_channel_queryoption callback to read the active
DSP setting of the faxdetect option.
* Made actually disable the active faxdetect DSP setting for the current
call on the analog port. my_handle_dtmfup() is used for normal analog
ports. dahdi_handle_dtmfup() is the legacy code and is no longer used
unless in a radio mode.
(closes issue #18116)
Reported by: seandarcy
Patches:
issue18116_v1.8.patch uploaded by rmudgett (license 664)
Review: https://reviewboard.asterisk.org/r/972/
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r291507 | rmudgett | 2010-10-13 14:01:48 -0500 (Wed, 13 Oct 2010) | 18 lines
Merged revision 291504 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
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r291504 | rmudgett | 2010-10-13 13:30:21 -0500 (Wed, 13 Oct 2010) | 11 lines
Hold off ast_hangup() from destroying the ast_channel.
Must get the ast_channel lock before proceeding with release_chan() and
release_chan_early() to hold off ast_hangup() from destroying the
ast_channel.
Missed this change for -r291468.
JIRA ABE-2598
JIRA SWP-2317
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Add the extension and context of the calling channel to the log output if a macro could not be found.
(closes issue #18112)
Reported by: prado
Patches:
app_macro-info.diff uploaded by prado (license 510)
Tested by: schmidts
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r291227 | dvossel | 2010-10-12 10:58:56 -0500 (Tue, 12 Oct 2010) | 16 lines
Fixes manager.c crash.
This issue was caused by improper use of the mansession lock and
manession_session lock. These two structures are confusing to begin
with so I'm not surprised this occurred. I fixed this by consistently
making sure we use each of these locks only to protect the data
in the corresponding structure. We had mismatched usage of these
locks which resulted in no mutual exclusivity occurring at all.
(closes issue #17994)
Reported by: vrban
Patches:
mansession_locking_fix.diff uploaded by dvossel (license 671)
Tested by: vrban
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r291192 | dvossel | 2010-10-11 16:38:39 -0500 (Mon, 11 Oct 2010) | 19 lines
Gtalk enhancements and general code cleanup.
This patch includes several chan_gtalk enhancements.
Two new gtalk.conf options have been added, externip
and stunadd. Setting externip allows us to
manually specify what the external IP address is
outside of a NAT environment. Setting the stunaddr
option to a valid stun server allows for that external
ip to be retrieved via a STUN server automatically. This
external IP is then advertised during call setup as
a possible candidate.
I have also attempted to clean up chan_gtalk's code
so it meets our coding guidelines. During this cleanup
I noticed several things that need to be done in the
code and made a TODO section at the top of the file.
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r291075 | rmudgett | 2010-10-11 11:42:54 -0500 (Mon, 11 Oct 2010) | 22 lines
Merged revisions 291073 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r291073 | rmudgett | 2010-10-11 11:39:17 -0500 (Mon, 11 Oct 2010) | 15 lines
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.
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r290973 | dvossel | 2010-10-08 15:44:59 -0500 (Fri, 08 Oct 2010) | 12 lines
Make outbound Google Voice calls.
This patch allows for outbound Google Voice calls to be
dialed from Asterisk using chan_gtalk. Below is an example
dialstring.
exten -> blah,1,Dial(Gtalk/asterisk/+15552225555@voice.google.com,,)
In this example, 'asterisk' is the jabber.conf profile configured
to connect to your gmail account. In order to receive Google Voice
calls make sure to enable 'allowguest=yes' in gtalk.conf.
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r290864 | jpeeler | 2010-10-07 21:56:24 -0500 (Thu, 07 Oct 2010) | 23 lines
Merged revisions 290863 via svnmerge from
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r290863 | jpeeler | 2010-10-07 21:45:44 -0500 (Thu, 07 Oct 2010) | 16 lines
Merged revisions 290862 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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r290829 | dvossel | 2010-10-07 17:38:05 -0500 (Thu, 07 Oct 2010) | 6 lines
Add Philippe Sultan to chan_gtalk author list.
Philippe has made some notable contributions to the
gtalk channel driver. His name deserves to be listed
amoung the authors of that file. Thanks Philippe!
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r290648 | dvossel | 2010-10-06 16:08:19 -0500 (Wed, 06 Oct 2010) | 12 lines
Fixes gtalk outbound DTMF to work properly.
Outbound DTMF with gtalk needs to be done within the RTP stream. I discovered
this after investigating a packet capture from the gmail client. Instead of
performing jingle signaling DTMF, the gtalk servers expect all DTMF to arrive
on the RTP stream using RFC2833 way of doing things. Chan_gtalk also had an issue
with negotiating RTP payload type 106 for the telephony-event and then sending
DTMF as payload 101. This has been resolved by always negotiating 101 as the payload
type like we do everywhere else. With this patch, incoming google voice calls forwarded
to Asterisk via gtalk work.
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r290542 | twilson | 2010-10-05 21:35:51 -0700 (Tue, 05 Oct 2010) | 6 lines
Don't try to send RTP when remote_address is null
It is possible for ast_rtp_stop() to be called which will clear the remote
address and cause the sendto to fail and spam warnings. Don't send in this
case.
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r290255 | tilghman | 2010-10-04 18:23:11 -0500 (Mon, 04 Oct 2010) | 18 lines
Merged revisions 290254 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r290254 | tilghman | 2010-10-04 18:14:59 -0500 (Mon, 04 Oct 2010) | 11 lines
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
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r289840 | jpeeler | 2010-10-01 21:43:45 -0500 (Fri, 01 Oct 2010) | 29 lines
Merged revisions 289798 via svnmerge from
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r289798 | jpeeler | 2010-10-01 18:01:31 -0500 (Fri, 01 Oct 2010) | 22 lines
Merged revisions 289797 via svnmerge from
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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
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r289701 | jpeeler | 2010-10-01 11:22:19 -0500 (Fri, 01 Oct 2010) | 28 lines
Merged revisions 289700 via svnmerge from
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r289700 | jpeeler | 2010-10-01 11:21:04 -0500 (Fri, 01 Oct 2010) | 21 lines
Merged revisions 289699 via svnmerge from
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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
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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.
Review: https://reviewboard.asterisk.org/r/901/
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r289549 | rmudgett | 2010-09-30 14:28:36 -0500 (Thu, 30 Sep 2010) | 17 lines
Merged revision 289547 from
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r289547 | rmudgett | 2010-09-30 14:16:36 -0500 (Thu, 30 Sep 2010) | 10 lines
In chan_misdn, the DivertingLegInformation2 DivertingNr is garbage when the number is restricted.
The same thing happens with DivertingLegInformation1 DivertedTo number.
The misdn_PresentedNumberUnscreened_extract() extracted the Unscreened
PartyNumber field unconditionally. It now checks the presented number
unscreened type to see if the PartyNumber was even present.
JIRA ABE-2595
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r289426 | russell | 2010-09-30 10:39:45 -0500 (Thu, 30 Sep 2010) | 22 lines
Merged revisions 289425 via svnmerge from
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r289425 | russell | 2010-09-30 10:37:29 -0500 (Thu, 30 Sep 2010) | 15 lines
Merged revisions 289424 via svnmerge from
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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
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r289340 | qwell | 2010-09-29 16:12:43 -0500 (Wed, 29 Sep 2010) | 22 lines
Merged revisions 289339 via svnmerge from
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r289339 | qwell | 2010-09-29 16:03:47 -0500 (Wed, 29 Sep 2010) | 15 lines
Merged revisions 289338 via svnmerge from
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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
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r289333 | russell | 2010-09-29 15:20:23 -0500 (Wed, 29 Sep 2010) | 11 lines
Merged revisions 289332 via svnmerge from
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r289332 | russell | 2010-09-29 15:15:57 -0500 (Wed, 29 Sep 2010) | 4 lines
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.
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r289099 | bbryant | 2010-09-28 14:18:02 -0400 (Tue, 28 Sep 2010) | 28 lines
Merged revisions 289095 via svnmerge from
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r289095 | bbryant | 2010-09-28 14:14:19 -0400 (Tue, 28 Sep 2010) | 21 lines
Merged revisions 289094 via svnmerge from
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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/
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r288748 | twilson | 2010-09-24 09:02:27 -0700 (Fri, 24 Sep 2010) | 19 lines
Merged revisions 288747 via svnmerge from
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r288747 | twilson | 2010-09-24 08:37:39 -0700 (Fri, 24 Sep 2010) | 12 lines
Merged revisions 288746 via svnmerge from
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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.
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r288341 | russell | 2010-09-22 11:45:18 -0500 (Wed, 22 Sep 2010) | 25 lines
Merged revisions 288340 via svnmerge from
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r288340 | russell | 2010-09-22 11:44:13 -0500 (Wed, 22 Sep 2010) | 18 lines
Merged revisions 288339 via svnmerge from
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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!
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r288194 | rmudgett | 2010-09-21 19:06:21 -0500 (Tue, 21 Sep 2010) | 40 lines
Merged revisions 288193 via svnmerge from
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r288193 | rmudgett | 2010-09-21 19:03:37 -0500 (Tue, 21 Sep 2010) | 33 lines
Merged revisions 288192 via svnmerge from
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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/
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r288159 | tilghman | 2010-09-21 17:57:22 -0500 (Tue, 21 Sep 2010) | 29 lines
Merged revisions 288113 via svnmerge from
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r288113 | tilghman | 2010-09-21 16:59:46 -0500 (Tue, 21 Sep 2010) | 22 lines
Merged revisions 288112 via svnmerge from
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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
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r288079 | rmudgett | 2010-09-21 15:29:51 -0500 (Tue, 21 Sep 2010) | 2 lines
Protect channel access in CONNECTED_LINE and REDIRECTING interception macro launch code.
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r288080 | rmudgett | 2010-09-21 15:29:59 -0500 (Tue, 21 Sep 2010) | 8 lines
Simplify locking code for REDIRECTING interception macro when forwarding a call.
Simplified the locking code by using a local copy of the redirecting party
information in app_dial.c:do_forward() and app_queue.c:wait_for_answer()
for launching the REDIRECTING interception macro when a call is forwarded.
Reduced the lock time of the 'o->chan' and 'in' channels.
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adding two dialog container, one for dialogs which need destroy, another for rtptimeout checks.
both container will be checked on every loop of do_monitor instead of iterate through all dialogs.
(closes issue #17912)
Reported by: schmidts
Tested by: schmidts
Review: https://reviewboard.asterisk.org/r/917/
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r287760 | bbryant | 2010-09-20 20:00:23 -0400 (Mon, 20 Sep 2010) | 30 lines
Merged revisions 287759 via svnmerge from
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r287759 | bbryant | 2010-09-20 19:58:26 -0400 (Mon, 20 Sep 2010) | 23 lines
Merged revisions 287758 via svnmerge from
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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]
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r287685 | alecdavis | 2010-09-21 11:16:45 +1200 (Tue, 21 Sep 2010) | 18 lines
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
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r287683 | rmudgett | 2010-09-20 18:14:42 -0500 (Mon, 20 Sep 2010) | 9 lines
The inalarm flag was not set in sig_analog struct if the port is initially in alarm.
Fixed initial inalarm value for sig_analog ports.
Along with -r261007, this gets the inalarm flag in sync with chan_dahdi
for sig_analog ports.
(closes issue #16983)
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r287661 | alecdavis | 2010-09-21 10:21:50 +1200 (Tue, 21 Sep 2010) | 14 lines
ast_do_masquerade. Keep channels ao2_container locked while unlink and linking channels.
Previously, Masquerade would unlock 'original' and 'clonechan' and allow another masq thread to run.
End result would be corrupted memory, and the frequent report 'Bad Magic Number'.
(closes issue #17801,#17710)
Reported by: notthematrix
Patches:
Based on bug17801.diff1.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
Review: https://reviewboard.asterisk.org/r/928
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r287647 | dvossel | 2010-09-20 17:09:16 -0500 (Mon, 20 Sep 2010) | 21 lines
Addition of the FrameHook API (AKA AwesomeHooks)
So far all our tools for viewing and manipulating media streams
within Asterisk have been entirely focused on audio. That made
sense then, but is not scalable now. The FrameHook API lets us
tap into and manipulate _ANY_ type of media or signaling passed
on a channel present today or in the future. This tool is a step
in the direction of expanding Asterisk's boundaries and will help
generate some rather interesting applications in the future.
In addition to the FrameHook API, a simple dialplan function
exercising the api has been included as well. This function
is called FRAME_TRACE(). FRAME_TRACE() allows for the internal
ast_frames read and written to a channel to be output. Filters
can be placed on this function to debug only certain types of frames.
This function could be thought of as an internal way of doing
ast_frame packet captures.
Review: https://reviewboard.asterisk.org/r/925/
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r287269 | pitel | 2010-09-17 10:37:49 +0200 (Pá, 17 zář 2010) | 8 lines
Support for HTTP redirects in calendar's URL
libneon does not support HTTP redirects (3xx responses) by default. You must tell it to follow them.
Also, another little unsigned int fix.
(closes issue #17776)
Review: https://reviewboard.asterisk.org/r/921/
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r287270 | pitel | 2010-09-17 10:42:37 +0200 (Pá, 17 zář 2010) | 6 lines
Asterisk crashing because of double free when EWS request fails
The free is done later in code. I think ast_free() should have built in checks for double free.
(closes issue #17782)
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r287271 | pitel | 2010-09-17 10:44:28 +0200 (Pá, 17 zář 2010) | 6 lines
Events are visible after they were removed from EWS calendar
Because we must merge calendar even when it's empty.
(closes issue #17786)
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r287056 | twilson | 2010-09-15 17:17:17 -0500 (Wed, 15 Sep 2010) | 10 lines
Don't hang up a call on an SRTP unprotect failure
Also make it more obvious when there is an issue en/decrypting.
(closes issue #17563)
Reported by: Alexcr
Patches:
res_srtp.c.patch uploaded by sfritsch (license 1089)
Tested by: twilson
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r287017 | rmudgett | 2010-09-15 15:53:38 -0500 (Wed, 15 Sep 2010) | 65 lines
Merged revision 287014 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
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r287014 | rmudgett | 2010-09-15 15:32:24 -0500 (Wed, 15 Sep 2010) | 58 lines
The handling of call transfer signaling for mISDN PTMP is not fully implemented.
The handling of call transfer signaling for mISDN PTMP is not fully
implemented. The signaling of number updates with ISDN/DSS1 ECT
supplementary services (ETS 300 369-1) comes along with a notification
indicator IE and redirection number IE for PTMP. The implementation in
the current Asterisk mISDN channel unfortunately can handle these
information elements only in a NOTIFY message. These information elements
are also signaled in a FACILTY message with a RequestSubaddress facility,
when the subscriber is already in the active state (see 9.2.4 and 9.2.5 of
ETS 300 369-1).
**********
abe_2526_ast.patch
* Added support to handle the notification indicator IE and redirection
number IE with the RequestSubaddress facility.
* Made misdn_update_connected_line() send a NOTIFY message if Asterisk
originated the call and it is not connected yet.
* Made misdn_update_connected_line() send a FACILITY message if the call
is already connected.
This patch requires the presence of the associated mISDN patches to
compile. I had to enhance mISDN to allow the notification indicator IE
and the redirection number IE to be used with a FACILITY message. Earlier
versions of the Digium enhanced mISDN are no longer going to work.
**********
abe_2526_misdn.patch
* Made an incoming FACILITY message allow the presence of the notification
indicator IE and the redirection number IE.
**********
abe_2526_misdnuser_v3.patch
* Added support to send and receive a FACILITY message with the
notification indicator IE and the redirection number IE.
* Added the ability to send a NOTIFY message in PTMP/NT mode to all
responding subcalls in Q.931 states 6, 7, 8, 9, and 25.
**********
Patches:
abe_2526_ast.patch uploaded by rmudgett (license 664)
abe_2526_misdn.patch uploaded by rmudgett (license 664)
abe_2526_misdnuser_v3.patch uploaded by rmudgett (license 664)
Tested by: rmudgett and reporter
JIRA SWP-2146
JIRA ABE-2526
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r287015 | jpeeler | 2010-09-15 15:32:52 -0500 (Wed, 15 Sep 2010) | 21 lines
Merged revisions 286998 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r286998 | jpeeler | 2010-09-15 15:28:02 -0500 (Wed, 15 Sep 2010) | 14 lines
Merged revisions 286941 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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r286931 | jpeeler | 2010-09-15 14:22:15 -0500 (Wed, 15 Sep 2010) | 16 lines
Add parking extension for non-default parking lots.
This is a new feature that allows for parking to custom parking lots to be
accessed directly, rather than with channel variables or by changing the
default parking lot. The extension is set with the parkext option just as the
default parking lot is done. Also, the manager action has been updated to
optionally allow a specified parking lot.
(closes issue #14882)
Reported by: vmikhnevych
Patches:
patch_14882.txt uploaded by mnick (license 874)
modified by me
Review: https://reviewboard.asterisk.org/r/884/
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r286904 | rmudgett | 2010-09-15 13:28:05 -0500 (Wed, 15 Sep 2010) | 12 lines
Unable to originate calls using E&M over T1.
When originating a call from Unit Under Test to Reference Unit using E&M
RBS signaling mode, I get the following warning message: "Ring/Off-hook in
strange state 3 on channel 1".
Fixed the sig_analog outgoing flag. It was never set when sig_analog was
extracted from chan_dahdi.
JIRA SWP-2191
JIRA AST-408
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r286905 | rmudgett | 2010-09-15 13:29:21 -0500 (Wed, 15 Sep 2010) | 1 line
Simplify some code in sig_analog.
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r286868 | mnicholson | 2010-09-15 08:05:52 -0500 (Wed, 15 Sep 2010) | 16 lines
Set tohost to the domain specified in the configuration file instead of the IP address of the host we are calling.
This fixes a regression introduced in r274783.
(closes issue #17960)
Reported by: adriavidal
Patches:
sip-tohost-fix1.diff uploaded by mnicholson (license 96)
Tested by: mich, mnicholson, adriavidal
(closes issue #17676)
Reported by: outcast
Patches:
sip-tohost-fix1.diff uploaded by mnicholson (license 96)
Tested by: mnicholson
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r286682 | mnicholson | 2010-09-14 13:04:21 -0500 (Tue, 14 Sep 2010) | 21 lines
Merged revisions 286681 via svnmerge from
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r286681 | mnicholson | 2010-09-14 13:02:24 -0500 (Tue, 14 Sep 2010) | 14 lines
Merged revisions 286679 via svnmerge from
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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)
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r286617 | pitel | 2010-09-14 08:55:44 +0200 (Út, 14 zář 2010) | 7 lines
Merging events for Exchange web service doesn't work as expected, resulting in only one event in calendar
The solution is to use "global" counter of events, since we do new requests for every event and calendar sync after every request. So now we do sync only after last request.
(closes issue #17877)
Review: https://reviewboard.asterisk.org/r/916/
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r286112 | russell | 2010-09-10 15:31:58 -0500 (Fri, 10 Sep 2010) | 9 lines
Rate limit calls to fsync() to 1 per second after astdb updates.
Astdb was determined to be one of the most significant bottlenecks in SIP
registration processing. This patch improved the speed of an astdb load
test by 50000% (yes, Fifty-Thousand Percent). On this particular load test
setup, this doubled the number of SIP registrations the server could handle.
Review: https://reviewboard.asterisk.org/r/825/
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r286189 | twilson | 2010-09-10 17:04:53 -0500 (Fri, 10 Sep 2010) | 30 lines
Merged revisions 286115 via svnmerge from
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r286115 | twilson | 2010-09-10 15:35:25 -0500 (Fri, 10 Sep 2010) | 23 lines
Merged revisions 286059 via svnmerge from
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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/
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r286118 | rmudgett | 2010-09-10 15:55:37 -0500 (Fri, 10 Sep 2010) | 25 lines
Merged revisions 286116 via svnmerge from
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r286116 | rmudgett | 2010-09-10 15:42:44 -0500 (Fri, 10 Sep 2010) | 18 lines
Merged revisions 286113 via svnmerge from
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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.
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r285962 | tilghman | 2010-09-10 00:32:18 -0500 (Fri, 10 Sep 2010) | 13 lines
Merged revisions 285961 via svnmerge from
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r285961 | tilghman | 2010-09-10 00:31:31 -0500 (Fri, 10 Sep 2010) | 6 lines
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.
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r285931 | tilghman | 2010-09-09 20:25:50 -0500 (Thu, 09 Sep 2010) | 21 lines
Merged revisions 285930 via svnmerge from
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r285930 | tilghman | 2010-09-09 20:16:32 -0500 (Thu, 09 Sep 2010) | 14 lines
Merged revisions 285889 via svnmerge from
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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).
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r285564 | dvossel | 2010-09-08 16:48:37 -0500 (Wed, 08 Sep 2010) | 60 lines
Merged revisions 285563 via svnmerge from
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r285563 | dvossel | 2010-09-08 16:47:29 -0500 (Wed, 08 Sep 2010) | 54 lines
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
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r285533 | bbryant | 2010-09-08 16:58:43 -0400 (Wed, 08 Sep 2010) | 15 lines
Merged revisions 285532 via svnmerge from
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r285532 | bbryant | 2010-09-08 16:56:12 -0400 (Wed, 08 Sep 2010) | 8 lines
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
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r285006 | dvossel | 2010-09-03 17:21:50 -0500 (Fri, 03 Sep 2010) | 9 lines
Disables auth_options_request option by default.
The auth_options_request option was created to do authentication
on OPTIONS request just like INVITES are done. Since it has been
noted that some endpoints use OPTIONS requests as a way of qualifying
a peer and that a 401 authentication response could result in
interoperability issues, this option has been disabled by default.
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r284950 | dvossel | 2010-09-03 12:29:02 -0500 (Fri, 03 Sep 2010) | 14 lines
authenticate OPTIONS requests just like we would an INVITE
OPTIONS requests should be treated the same as an INVITE
This includes authentication. This patch adds the ability for
incoming out of dialog OPTION requests to be authenticated
before providing a response indicating whether an extension
is available or not. The authentication routine works the
exact same way as it does for incoming INVITEs. This means
that if a peer has 'insecure=invite' in their peer definition,
the same will be true for the processing of the OPTIONS request.
Review: https://reviewboard.asterisk.org/r/881/
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r284597 | tilghman | 2010-09-02 00:00:34 -0500 (Thu, 02 Sep 2010) | 29 lines
Merged revisions 284593,284595 via svnmerge from
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r284593 | tilghman | 2010-09-01 17:59:50 -0500 (Wed, 01 Sep 2010) | 18 lines
Merged revisions 284478 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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/
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r284595 | tilghman | 2010-09-01 22:57:43 -0500 (Wed, 01 Sep 2010) | 2 lines
Failed to rerun bootstrap.sh after last commit
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r284561 | dvossel | 2010-09-01 16:47:01 -0500 (Wed, 01 Sep 2010) | 9 lines
During request to dialog matching, verify init_ruri is present before comparing.
During request to dialog matching, we attempt a best effort routine for fork
detection which requires several elements to be in place. The dialog's
initial request uri is one of those elements. Since it is best effort,
if the init_ruri is not present for some reason we can not proceed with that
routine.
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r284477 | twilson | 2010-09-01 13:44:36 -0500 (Wed, 01 Sep 2010) | 17 lines
Fix SRTP for changing SSRC and multiple a=crypto SDP lines
Adding code to Asterisk that changed the SSRC during bridges and masquerades
broke SRTP functionality. Also broken was handling the situation where an
incoming INVITE had more than one crypto offer. This patch caches the SRTP
policies the we use so that we can change the ssrc and inform libsrtp of the
new streams. It also uses the first acceptable a=crypto line from the incoming
INVITE.
(closes issue #17563)
Reported by: Alexcr
Patches:
srtp.diff uploaded by twilson (license 396)
Tested by: twilson
Review: https://reviewboard.asterisk.org/r/878/
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r284281 | tilghman | 2010-08-30 17:28:47 -0500 (Mon, 30 Aug 2010) | 18 lines
Merged revisions 284280 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r284280 | tilghman | 2010-08-30 17:27:06 -0500 (Mon, 30 Aug 2010) | 11 lines
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
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You can't write "same as above" in hypertext documentation. Above doesn't make sense in
hyperspace.
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r284065 | russell | 2010-08-28 16:29:45 -0500 (Sat, 28 Aug 2010) | 13 lines
Be more flexible with whitespace on AMI action headers.
Previously, this code required exactly one space to be after the ':' in headers
for an AMI action. This now makes whitespace optional, and allows whitespace that
is there to vary in amount.
(closes issue #17862)
Reported by: cmoye
Patches:
manager.c.patch_trunk uploaded by cmoye (license 858)
manager.c.patch_1.8 uploaded by cmoye (license 858)
Tested by: cmoye
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r283692 | dvossel | 2010-08-26 10:26:37 -0500 (Thu, 26 Aug 2010) | 32 lines
Merged revisions 283691 via svnmerge from
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r283691 | dvossel | 2010-08-26 10:24:40 -0500 (Thu, 26 Aug 2010) | 25 lines
Merged revisions 283690 via svnmerge from
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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.
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r283595 | dvossel | 2010-08-25 17:57:56 -0500 (Wed, 25 Aug 2010) | 14 lines
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r283594 | dvossel | 2010-08-25 17:56:42 -0500 (Wed, 25 Aug 2010) | 7 lines
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.
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r283559 | dvossel | 2010-08-25 10:54:11 -0500 (Wed, 25 Aug 2010) | 16 lines
Merged revisions 283558 via svnmerge from
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r283558 | dvossel | 2010-08-25 10:52:54 -0500 (Wed, 25 Aug 2010) | 10 lines
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)
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r283382 | dvossel | 2010-08-24 11:11:18 -0500 (Tue, 24 Aug 2010) | 25 lines
Merged revisions 283381 via svnmerge from
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r283381 | dvossel | 2010-08-24 11:07:37 -0500 (Tue, 24 Aug 2010) | 18 lines
Merged revisions 283380 via svnmerge from
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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.
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r283230 | russell | 2010-08-23 08:23:12 -0500 (Mon, 23 Aug 2010) | 7 lines
Make the AST_CEL_AMA enum match up with the AST_CDR_ ama flag values.
Really, having 2 enums for this is silly and error prone, demonstrated by
the crash that I hit because there was an assumption in the code that the
values in each matched up. However, this is a quick fix to get them to
match up so it will work.
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r283173 | russell | 2010-08-23 06:58:34 -0500 (Mon, 23 Aug 2010) | 5 lines
Expand cel_custom.conf.sample.
Include the usage of CSV_QUOTE() to ensure data has valid CSV formatting. Also list
the special CEL variables that are available for use in the mapping.
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r283050 | rmudgett | 2010-08-20 10:35:38 -0500 (Fri, 20 Aug 2010) | 36 lines
Merged revisions 283049 via svnmerge from
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r283049 | rmudgett | 2010-08-20 10:31:03 -0500 (Fri, 20 Aug 2010) | 29 lines
Merged revisions 283048 via svnmerge from
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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)
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r282895 | dvossel | 2010-08-19 16:07:20 -0500 (Thu, 19 Aug 2010) | 25 lines
Merged revisions 282894 via svnmerge from
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r282894 | dvossel | 2010-08-19 16:05:54 -0500 (Thu, 19 Aug 2010) | 18 lines
Merged revisions 282893 via svnmerge from
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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
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Consolidates all offhook (new call with dialtone) to setsubstate_offhook. This should be roughly equivalent to existing code, although a couple of calls now run through the full offhook sequence rather than an abbreviated one.
(closes issue #17812)
Reported by: wedhorn
Patches:
cleanup.stateoffhook.diff uploaded by wedhorn (license 30)
Tested by: salecha, wedhorn
Review: NA
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r282671 | rmudgett | 2010-08-18 10:27:51 -0500 (Wed, 18 Aug 2010) | 1 line
Use the correct operator when calculating the PRI span devstate.
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r282672 | rmudgett | 2010-08-18 10:28:27 -0500 (Wed, 18 Aug 2010) | 1 line
Use the correct type for aoce_delayhangup bit field.
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r282639 | mnicholson | 2010-08-18 08:10:39 -0500 (Wed, 18 Aug 2010) | 13 lines
Properly handle 200 and unknown responses conatined in NOTIFY requests received in response to REFER requests.
This patch fixes the way asterisk handles NOTIFY requests received in response to REFER requests. These changes to NOTIFY handler were first introduced in r217482. This new change properly handles the 200 response by queueing an AST_TRANSFER_SUCCESS control frame and also prevents that control frame from being queued when provisional and unknown responses are received.
(issue #17486)
Reported by: davidw
Tested by: mnicholson
(issue #12713)
Reported by: davidw
Review: https://reviewboard.asterisk.org/r/860/
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r282468 | twilson | 2010-08-16 12:53:44 -0500 (Mon, 16 Aug 2010) | 30 lines
Merged revisions 282467 via svnmerge from
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r282467 | twilson | 2010-08-16 12:32:01 -0500 (Mon, 16 Aug 2010) | 23 lines
Merged revisions 282430 via svnmerge from
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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/
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r282470 | lmadsen | 2010-08-16 13:01:00 -0500 (Mon, 16 Aug 2010) | 15 lines
Merged revisions 282469 via svnmerge from
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r282469 | lmadsen | 2010-08-16 13:00:09 -0500 (Mon, 16 Aug 2010) | 7 lines
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
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r282334 | rmudgett | 2010-08-13 18:53:36 -0500 (Fri, 13 Aug 2010) | 6 lines
PRI CCSS may use a stale dial string for the recall dial string.
If an outgoing call negotiates a different B channel than initially
requested, the saved original dial string was not transferred to the new B
channel. CCSS uses that dial string to generate the recall dial string.
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r282236 | dvossel | 2010-08-13 13:58:10 -0500 (Fri, 13 Aug 2010) | 23 lines
Merged revisions 282235 via svnmerge from
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r282235 | dvossel | 2010-08-13 13:54:53 -0500 (Fri, 13 Aug 2010) | 16 lines
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
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r282200 | twilson | 2010-08-13 11:00:02 -0500 (Fri, 13 Aug 2010) | 10 lines
Detect when libsrtp cannot be linked in a shared library
The libsrtp build system currently does not produce a shared library
or a static library compiled with -fPIC, so on 64-bit systems it is
possible that we will get a compile error if libsrtp is installed and
res_srtp is selected in menuselect.
This patch attempts to detect this situation and provide the user with
instructions to work around the problem.
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r282201 | twilson | 2010-08-13 11:02:20 -0500 (Fri, 13 Aug 2010) | 2 lines
Whitespace fix :-/
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r282098 | rmudgett | 2010-08-12 17:06:06 -0500 (Thu, 12 Aug 2010) | 7 lines
Separate call completion config parameter allocation and default initialization.
If you ever have a need to reset the call completion config parameters
to defaults, now you can.
And no Virginia, C++ idioms do not always work in C.
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r282047 | dvossel | 2010-08-12 15:15:41 -0500 (Thu, 12 Aug 2010) | 35 lines
improved translation paths for wideband codecs
The problem I'm addressing is that Asterisk's current
method of building the least cost translation paths
between codecs does not take into account sample rate.
For instance, it was possible for siren14 (a 32khz codec),
to contain the a translation path to siren7 (a 16khz
audio codec) that goes through slin at 8khz. In this
case Asterisk takes a 32khz codec, down samples it to
8khz and then up samples it to 16khz which is terrible
regardless if it is computationally less expensive. This
patch now builds translation paths that give priority to
maintaining the best possible sample rate before taking
into consideration computational cost. This patch also
adds cli commands to expose what translation paths are
actually being used.
Changes:
1. Translation paths will never contain a step that changes
the sample rate unless absolutely necessary.
2. When choosing the best codec to make two channels compatible.
Shared codecs with the highest sample rate are given priority.
3. A new cli command to show all translation paths available
for a specific codec 'core show translation paths [codec name]'
has been added.
4. 'core show translation' which displays the translation
matrix now includes the new higher bit audio codecs in the table.
5. 'core show channel [channel name]' now displays the
translation paths if translation is used.
(closes issue #16841)
Reported by: dvossel
Review: https://reviewboard.asterisk.org/r/842/
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r281982 | russell | 2010-08-12 11:33:30 -0500 (Thu, 12 Aug 2010) | 5 lines
Remove debugging output from verbose messages.
Pointer values to internal objects is not terribly useful to users in the
verbose messages about adding extensions and contexts.
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r281913 | jpeeler | 2010-08-11 22:03:37 -0500 (Wed, 11 Aug 2010) | 34 lines
Merged revisions 281912 via svnmerge from
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r281912 | jpeeler | 2010-08-11 22:01:38 -0500 (Wed, 11 Aug 2010) | 27 lines
Merged revisions 281911 via svnmerge from
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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
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r281575 | russell | 2010-08-10 13:05:07 -0500 (Tue, 10 Aug 2010) | 16 lines
Merged revisions 281574 via svnmerge from
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r281574 | russell | 2010-08-10 13:04:32 -0500 (Tue, 10 Aug 2010) | 9 lines
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)
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r281432 | dvossel | 2010-08-09 15:47:53 -0500 (Mon, 09 Aug 2010) | 20 lines
Merged revisions 281430 via svnmerge from
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r281430 | dvossel | 2010-08-09 15:46:50 -0500 (Mon, 09 Aug 2010) | 13 lines
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)
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r281429 | jpeeler | 2010-08-09 15:43:54 -0500 (Mon, 09 Aug 2010) | 27 lines
Merged revisions 281391 via svnmerge from
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r281391 | jpeeler | 2010-08-09 15:07:29 -0500 (Mon, 09 Aug 2010) | 20 lines
Merged revisions 281390 via svnmerge from
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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/
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r281294 | russell | 2010-08-09 07:14:34 -0500 (Mon, 09 Aug 2010) | 5 lines
Reorder some options in cdr.conf.sample.
Put all of the options that affect the contents of CDRs together, instead
of having the batch mode options in the middle of them.
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Cleaned up handling of onhook indications and added indications if more than one sub on device. Also fixes issue in 12324 so that the phone can call itself without locking up.
(closes issue #17692)
Reported by: jmhunter
Patches:
chan_skinny-transfer-v4.txt uploaded by DEA (license 3)
skinnytransfver.v8.diff uploaded by wedhorn (license 30)
Tested by: jmhunter, salecha, wedhorn
Review: NA
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@281257 f38db490-d61c-443f-a65b-d21fe96a405b
Move call answering stuff into new setsubstate_connected. Also add sub->substate var and set it to SUBSTATE_CONNECTED in setsubstate_connected.
(closes issue #17772)
Reported by: wedhorn
Patches:
cleanup.stateconnected2.diff uploaded by wedhorn (license 30)
Tested by: wedhorn, salecha
Review: NA
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@281227 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r281052 | russell | 2010-08-05 08:16:11 -0500 (Thu, 05 Aug 2010) | 16 lines
Merged revisions 281051 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r281051 | russell | 2010-08-05 08:11:32 -0500 (Thu, 05 Aug 2010) | 9 lines
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.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@281054 f38db490-d61c-443f-a65b-d21fe96a405b
Moved inline packet sending to transmit_ subs. Removed handle_keep_alive and handle_register_message to inline in handle_message. Also moved transmit_response(d) to transmit_response_bysessions(s) and created a wrapper transmit_response(d) that calls transmit_response_bysession(d->session).
(closes issue #16980)
Reported by: wedhorn
Patches:
skinny-clean06b.diff uploaded by wedhorn (license 30)
Tested by: wedhorn, DEA
Review: NA
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@280589 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r280307 | mnicholson | 2010-07-29 08:56:35 -0500 (Thu, 29 Jul 2010) | 11 lines
Merged revisions 280306 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r280306 | mnicholson | 2010-07-29 08:45:11 -0500 (Thu, 29 Jul 2010) | 2 lines
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/
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Additionally, pass AST_CONTROL_T38_PARAMETERS control frames through generic bridges. This change appears to have been unintentionally left out of rev 203699.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@280308 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r279953 | russell | 2010-07-27 16:16:05 -0500 (Tue, 27 Jul 2010) | 5 lines
Add --enable-coverage option to configure script.
This option enables the proper compiler flags for tracking code coverage, which
is useful along side automated testing.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@279954 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r279949 | dvossel | 2010-07-27 15:57:00 -0500 (Tue, 27 Jul 2010) | 31 lines
Merged revisions 279946 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r279946 | dvossel | 2010-07-27 15:54:32 -0500 (Tue, 27 Jul 2010) | 24 lines
Merged revisions 279945 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@279951 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r279916 | russell | 2010-07-27 14:50:56 -0500 (Tue, 27 Jul 2010) | 12 lines
Fix inband DTMF detection on outgoing ISDN calls.
This is a regression from the sig_pri split from chan_dahdi. When a call is
first initiated, the inband DTMF detector is not enabled if it's an outgoing
ISDN call. However, it needs to be turned on once the media path starts up.
This handling was put back in the open_media() callback of chan_dahdi. In
sig_pri, open_media() calls were added to a few places where it was needed,
including handling of PRI_EVENT_RINGING, PRI_EVENT_PROGRESS, and
PRI_EVENT_PROCEEDING.
Thanks to rmudgett for helping me with the patch!
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@279917 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r279887 | mmichelson | 2010-07-27 13:54:07 -0500 (Tue, 27 Jul 2010) | 16 lines
Fix parsing error in sip_sipredirect().
The code was written in a way that did a bad job of
parsing the port out of a URI. Specifically, it would
do badly when dealing with an IPv6 address. In this
particular scenario, there was no value from parsing
the port out, so I just removed that logic. And while
I was messing around in the function, I changed some
variable names to be more descriptive.
(closes issue #17661)
Reported by: oej
Patches:
17661.diff uploaded by mmichelson (license 60)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@279888 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r279785 | mmichelson | 2010-07-27 10:15:22 -0500 (Tue, 27 Jul 2010) | 20 lines
Merged revisions 279784 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r279784 | mmichelson | 2010-07-27 10:13:24 -0500 (Tue, 27 Jul 2010) | 14 lines
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
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@279786 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r279658 | qwell | 2010-07-26 18:03:38 -0500 (Mon, 26 Jul 2010) | 12 lines
Merged revisions 279657 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r279657 | qwell | 2010-07-26 17:59:52 -0500 (Mon, 26 Jul 2010) | 5 lines
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.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@279659 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r279568 | dvossel | 2010-07-26 14:59:03 -0500 (Mon, 26 Jul 2010) | 21 lines
transaction matching using top most Via header
This patch modifies the way chan_sip.c does transaction to dialog
matching. Asterisk now stores information in the top most Via header
of the initial incoming request and compares that against other Requests
that have the same call-id. This results in Asterisk being able to
detect a forked call in which it has received multiple legs of the
fork. I completely stripped out the previous matching code and made
the comparisons a little more explicit and easier to understand. My
comments in the code should offer all the details involving this patch.
This patch also fixes a bug with the usage of the OBJ-MULTIPLE flag to
find multiple dialogs with the same call-id. Since the callback
function was returning (CMP_MATCH | CMP_STOP) only the first item
found was being returned. I fixed this by making a new callback
function for finding multiple dialogs that only returns (CMP_MATCH)
on a match allowing for multiple items to be returned.
Review: https://reviewboard.asterisk.org/r/776/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@279569 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r279504 | mmichelson | 2010-07-26 11:04:09 -0500 (Mon, 26 Jul 2010) | 14 lines
Allow for systems without locale support to be usable.
A recent change to SIP URI comparison code added a locale-specific
string comparison to the mix, and certain systems do not support
such functions. This fix allows for those systems to still use
Asterisk 1.8
(closes issue #17697)
Reported by: pprindeville
Patches:
asterisk-trunk-bugid17697.patch uploaded by pprindeville (license 347)
Tested by: mmichelson
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@279533 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r279227 | rmudgett | 2010-07-23 17:20:47 -0500 (Fri, 23 Jul 2010) | 21 lines
Merged revisions 279207 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r279207 | rmudgett | 2010-07-23 17:11:23 -0500 (Fri, 23 Jul 2010) | 14 lines
Merged revisions 279206 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@279245 f38db490-d61c-443f-a65b-d21fe96a405b
The "dahdi show channels" extension column previously only showed the
called number of an incoming call. It now shows the called number for an
outgoing call as well.
(closes issue #17653)
Reported by: amazinzay
Patches:
issue17653_trunk.txt uploaded by rmudgett (license 664)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@279156 f38db490-d61c-443f-a65b-d21fe96a405b
res=ast_play_and_record(chan,"priv-recordintro",pa->privintro,4,"gsm",&duration,silencethreshold,2000,0);/* NOTE: I've reduced the total time to 4 sec */
res=ast_play_and_record(chan,"priv-recordintro",pa->privintro,4,"sln",&duration,silencethreshold,2000,0);/* NOTE: I've reduced the total time to 4 sec */
/* don't think we'll need a lock removed, we took care of