(closes issue #16263)
Reported by: andrew
Patches:
pagerdate.patch uploaded by andrew (license 240)
(with a slight modification by me)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@232916 f38db490-d61c-443f-a65b-d21fe96a405b
Previously only possible per context, new option called imapfolder.
(closes issue #14298)
Reported by: jablko
Patches:
patch-200906202 uploaded by jablko (license 675)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@232700 f38db490-d61c-443f-a65b-d21fe96a405b
Internally the way T.38 is negotiated has changed and the option no longer
reflects a behavior that is valid. It will now look for a CNG tone on
received calls and if present send the call to the 'fax' extension. It is
then up to the application or channel to request the switch over to T.38.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@230881 f38db490-d61c-443f-a65b-d21fe96a405b
Just updating a spelling error and some capitalization in a
documentation update that Olle added. May the Swenglish be
with you.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@229639 f38db490-d61c-443f-a65b-d21fe96a405b
Update the extensions.conf.sample [stdexten] context so that we use the
variable instead of requiring it to be passed explicitly. Also updated uses of
the [stdexten] context throughout.
(closes issue #15858)
Reported by: pprindeville
Patches:
stdexten-context-update.txt uploaded by lmadsen (license 10)
Tested by: pprindeville
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@227361 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r226382 | lmadsen | 2009-10-28 15:06:13 -0500 (Wed, 28 Oct 2009) | 9 lines
Update documentation in sip.conf.sample.
Update the documentation in sip.conf.sample in order to make it more clear
that directmedia/canreinvite do not cause Asterisk to ignore reINVITEs. It
is only used to stop Asterisk from generating a reINVITE, but does not stop
it from accepting them if necessary.
(closes issue #15644)
Reported by: lmadsen
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@226384 f38db490-d61c-443f-a65b-d21fe96a405b
This change adds a configuration option to SIP peers, unsolicited_mailbox, which
configures a virtual mailbox to use for received new/old MWI information. This
virtual mailbox can then be used by any device supporting MWI.
(closes issue #13028)
Reported by: AsteriskRocks
Patches:
bug_13028_chan_sip_external_mwi_20090707.patch uploaded by cmaj (license 830)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@226060 f38db490-d61c-443f-a65b-d21fe96a405b
* Added handling of received HOLD/RETRIEVE messages and the optional ability
to transfer a held call on disconnect similar to an analog phone.
* Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
Will reroute/deflect an outgoing call when receive the message.
Can use the DAHDISendCallreroutingFacility to send the message for the
supported switches.
* Added ability to send/receive keypad digits in the SETUP message.
Send keypad digits in SETUP message: Dial(DAHDI/g1[/K<keypad_digits>][/extension])
Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
* Added support for BRI PTMP NT mode.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@225692 f38db490-d61c-443f-a65b-d21fe96a405b
This is the second commit for this and documents the text stream using the configured
IP address and fixes a bug in the original patch where the UDPTL stream would also
use the different IP address.
(closes issue #14729)
Reported by: _brent_
Patches:
media_address.patch uploaded by brent (license 388)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@225089 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r225032 | dvossel | 2009-10-21 09:37:04 -0500 (Wed, 21 Oct 2009) | 20 lines
IAX/SIP shrinkcallerid option
The shrinking of caller id removes '(', ' ', ')', non-trailing '.',
and '-' from the string. This means values such as 555.5555 and
test-test result in 555555 and testtest. There are instances,
such as Skype integration, where a specific value is passed via
caller id that must be preserved unmodified. This patch makes
the shrinking of caller id optional in chan_sip and chan_iax in
order to support such cases. By default this option is on to
preserve previous expected behavior.
(closes issue #15940)
Reported by: dimas
Patches:
v2-15940.patch uploaded by dimas (license 88)
15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671)
Tested by: dvossel
Review: https://reviewboard.asterisk.org/r/408/
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@225033 f38db490-d61c-443f-a65b-d21fe96a405b
Explains new options for detecting DTMF CID on fxo lines
(issue #9096)
Reported by: fleed
Patches:
chan_dahid_sample_config.patch uploaded by sum (license 766)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@224144 f38db490-d61c-443f-a65b-d21fe96a405b
Ability has been added to both manager command SIPnotify as well as console
command sip notify. Message body is stored in the "Content" variable. An
example is present in sip_notify.conf.
(closes issue #13926)
Reported by: jthurman
Patches:
sip-notify-svn189463.diff uploaded by gareth (license 208)
Tested by: gareth
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@224035 f38db490-d61c-443f-a65b-d21fe96a405b
This patch adds support for TCP/TLS in the Contact header when using
NAT, specifically externip or externhost. The original issue was that
Asterisk sent 5060 as the port in the contact header whether TLS was
used or not. Additionally, this patch adds 2 config options to sip.conf,
specifically externtcpport and externtlsport. This allows a user to
specify different external ports for TCP and TLS other than those used
internally, this is especially useful in in a PAT/port redirection setup.
Thanks to ebroad for reporting the issue and providing the patch!
(closes issue #15880)
Reported by: ebroad
Patches:
portmap.patch uploaded by ebroad (license 878)
externtXXport_v2.patch uploaded by ebroad (license 878)
Tested by: ebroad
Review: https://reviewboard.asterisk.org/r/392/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@222398 f38db490-d61c-443f-a65b-d21fe96a405b
Many T.38 endpoints incorrectly send the maximum IFP frame size they can accept
as the T38FaxMaxDatagram value in their SDP, when in fact this value is
supposed to be the maximum UDPTL payload size (datagram size) they can accept.
If the value they supply is small enough (a commonly supplied value is '72'),
T.38 UDPTL transmissions will likely fail completely because the UDPTL packets
will not have enough room for a primary IFP frame and the redundancy used for
error correction. If this occurs, the Asterisk UDPTL stack will emit log messages
warning that data loss may occur, and that the value may need to be overridden.
This patch extends the 't38pt_udptl' configuration option in sip.conf to allow
the administrator to override the value supplied by the remote endpoint and
supply a value that allows T.38 FAX transmissions to be successful with that
endpoint. In addition, in any SIP call where the override takes effect, a debug
message will be printed to that effect. This patch also removes the
T38FaxMaxDatagram configuration option from udptl.conf.sample, since it has not
actually had any effect for a number of releases.
In addition, this patch cleans up the T.38 documentation in sip.conf.sample
(which incorrectly documented that T.38 support was passthrough only).
(issue #15586)
Reported by: globalnetinc
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@222110 f38db490-d61c-443f-a65b-d21fe96a405b
chan_sip has had the ability to control T.38 FAX error correction mode on a per-peer
(or global) basis for a couple of releases now, which is where it should have been
all along. This patch removes the ability to configure it in udptl.conf, but issues
a warning if the user tries to do, telling them to look at sip.conf.sample for how
to configure it now. For any SIP peers that are T.38 enabled in sip.conf, there is
already a default for FEC error correction even if the user does not specify any mode,
so this change will not turn off error correction by default, it will have the same
default value that has been in the udptl.conf sample file.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@221592 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r221360 | mnicholson | 2009-09-30 14:36:06 -0500 (Wed, 30 Sep 2009) | 10 lines
Fix SRV lookup and Request-URI generation in chan_sip.
This patch adds a new field "portinuri" to the sip dialog struct and the sip peer struct. That field is used during RURI generation to determine if the port should be included in the RURI. It is also used in some places to determine if an SRV lookup should occur.
(closes issue #14418)
Reported by: klaus3000
Tested by: klaus3000, mnicholson
Review: https://reviewboard.asterisk.org/r/369/
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@221432 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009) | 25 lines
Change the SSRC by default when our media stream changes
Be default, change SSRC when doing an audio stream changes Asterisk doesn't
honor marker bit when reinvited to already-bridged RTP streams,resulting in
far-end stack discarding packets with "old" timestamps that areactually part of
a new stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is a
reinvite, unless the 'constantssrc' is set to true in sip.conf.
The original issue reported to Digium support detailed the following situation:
ITSP <-> Asterisk 1.4.26.2 <-> SIP-based Application Server Call comes in
fromITSP, Asterisk dials the app server which sends a re-invite back
toAsterisk--not to negotiate to send media directly to the ITSP, but to
indicatethat it's changing the stream it's sending to Asterisk. The app
servergenerates a new SSRC, sequence numbers, timestamps, and sets the marker
bit on the new stream. Asterisk passes through the teimstamp of the new stream,
butdoes not reset the SSRC, sequence numbers, or set the marker bit.
When the timestamp on the new stream is older than the timestamp on the
originalstream, the ITSP (which doesn't know there has been any change) discards
the newframes because it thinks they are too old. This patch addresses this by
changing the SSRC on a stream update unless constantssrc=true is set in
sip.conf.
Review: https://reviewboard.asterisk.org/r/374/
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@221266 f38db490-d61c-443f-a65b-d21fe96a405b
JABBER_RECEIVE (along with JabberSend) makes Asterisk interact with users over
XMPP to process calls.
SendText can be used instead of JabberSend in the context of XMPP based voice
channels (chan_gtalk and chan_jingle).
(closes issue #12569)
Reported by: eech55
Tested by: phsultan, asannucci, lmadsen, jtodd, maxgo
Review: https://reviewboard.asterisk.org/r/88/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@220457 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27 lines
Make apps send PROGRESS control frame for early media and fix too early media issue in SIP
The issue at hand is that some legacy (dying) PBX systems send empty media frames on PRI
links *before* any call progress. The SIP channel receives these frames and by default
signals 183 Session progress and starts sending media. This will cause phones to
play silence and ignore the later 180 ringing message. A bad user experience.
The fix is twofold:
- We discovered that asterisk apps that support early media ("noanswer") did not send
any PROGRESS frame to indicate early media. Fixed.
- We introduce a setting in chan_sip so that users can disable any relay of media frames
before the outbound channel actually indicates any sort of call progress.
In 1.4, 1.6.0 and 1.6.1, this will be disabled for backward compatibility. In later versions
of Asterisk, this will be enabled. We don't assume that it will change your Asterisk
phone experience - only for the better.
We encourage third-party application developers to make sure that if they have applications
that wants to send early media, add a PROGRESS control frame transmission to make sure that
all channel drivers actually will start sending early media. This has not been the default
in Asterisk previous to this patch, so if you got inspiration from our code, you need to
update accordingly. Sorry for the trouble and thanks for your support.
This code has been running for a few months in a large scale installation (over 250
servers with PRI and/or BRI links to old PBX systems).
That's no proof that this is an excellent patch, but, well, it's tested :-)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@216438 f38db490-d61c-443f-a65b-d21fe96a405b
Allows characters that are otherwise used as delimiters to be used within
certain fields (like the secret).
(closes issue #15008, closes issue #15672)
Reported by: tilghman
Patches:
20090818__issue15008.diff.txt uploaded by tilghman (license 14)
Tested by: lmadsen, tilghman
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@213098 f38db490-d61c-443f-a65b-d21fe96a405b
This eliminates a future source of possible confusion with the configuration of
1.6.1 and higher.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@212857 f38db490-d61c-443f-a65b-d21fe96a405b
It is clear from multiple mailing list, forum, wiki and other sorts of posts
that users don't really understand the effects that the 'canreinvite' config
option actually has, and that in some cases they think that setting it to 'no'
will actually cause various other features (T.38, MOH, etc.) to not work properly,
when in fact this is not the case. This patch changes the proper name of the
option to what it should have been from the beginning ('directmedia'), but
preserves backwards compatibility for existing configurations.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@210190 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r209131 | mmichelson | 2009-07-27 12:44:06 -0500 (Mon, 27 Jul 2009) | 18 lines
Allow for UDPTL to use only even-numbered ports if desired.
There are some VoIP providers out there that will not accept SDP
offers with odd numbered UDPTL ports. While it is my personal opinion
that these VoIP providers are misinterpreting RFC 2327, it really is
not a big deal to play along with their silly little games. Of course,
since restricting UDPTL ports to only even numbers reduces the range
of available ports by half, so the option to use only even port numbers
is off by default. A user can enable the behavior by setting
use_even_ports=yes in udptl.conf.
(closes issue #15182)
Reported by: CGMChris
Patches:
15182.patch uploaded by mmichelson (license 60)
Tested by: CGMChris
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@209132 f38db490-d61c-443f-a65b-d21fe96a405b
Requiring 'module reload' to reload everything, including
core etc makes russell very unhappy.
The default configuration already loads the 'friendly' aliases template.
Added 'reload=module reload' to that template.
Also removed the comment in main/cli.c that reload should come back.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@208813 f38db490-d61c-443f-a65b-d21fe96a405b
This is a continuation of revision 885 to LibPRI (Capture and expose the Reverse
Charging Indication IE on ISDN PRI) which added the ability to get/set Reverse
Charging Indication in LibPRI. This patch adds the ability to specify RCI on
the outbound leg of a PRI call from within Asterisk, by prefixing the dialed
number with a capital 'C' like:
...,Dial(DAHDI/g1/C4445556666)
And to read it off an inbound channel:
exten => s,1,Set(RCI=${CHANNEL(reversecharge)})
Thanks again to rmudgett for the thorough review.
(closes issue #13760)
Reported by: mrgabu
Review: https://reviewboard.asterisk.org/r/303/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@204749 f38db490-d61c-443f-a65b-d21fe96a405b
Someone asked yesterday, "is there a good reason why we can't just put these
modules in Asterisk?". After a brief discussion, as long as the modules are
clearly set aside in their own directory and not enabled by default, it is
perfectly fine.
For more information about why a module goes in addons, see README-addons.txt.
chan_ooh323 does not currently compile as it is behind some trunk API updates.
However, it will not build by default, so it should be okay for now.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@204413 f38db490-d61c-443f-a65b-d21fe96a405b
This might seem like a legitimate comment that merely needed semicolon
prefixes, but in reality, the adaptive layer is designed to allow arbitrary
CDR variables, without needing the use of a userfield to store multiple items.
It's therefore not only invalid syntax but also goes against the intent of the
adaptive method.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@204069 f38db490-d61c-443f-a65b-d21fe96a405b
The original patch for this was written by Brett Bryant, and I split it out into
it's own module.
(closes issue #12876)
Reported by: bbryant
Patches:
06162008_cdr_custom_syslog.diff uploaded by bbryant (license 36)
05212009_cdr_syslog.patch uploaded by seanbright (license 71)
Tested by: seanbright
Review: https://reviewboard.asterisk.org/r/297/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@203846 f38db490-d61c-443f-a65b-d21fe96a405b
CEL is the new system for logging channel events. This was inspired after
facing many problems trying to represent what is possible to happen to a call
in Asterisk using CDR records. For more information on CEL, see the built in
HTML or PDF documentation generated from the files in doc/tex/.
Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard
work developing this code. Also, thanks to Matt Nicholson (mnicholson) and
Sean Bright (seanbright) for their assistance in the final push to get this
code ready for Asterisk trunk.
Review: https://reviewboard.asterisk.org/r/239/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@203638 f38db490-d61c-443f-a65b-d21fe96a405b
Also change the preferred configuration option from 'hostname' (which was
misleading because it didn't actually treat the value as a hostname) to
'connection' and added some verbage explaining that the user would need to
refer to their freetds.conf file for those settings. 'hostname' was kept
as a backwards compatible configuration parameter.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@202887 f38db490-d61c-443f-a65b-d21fe96a405b
chan_sip has an option to save the sysname on rtupdate. This patch copies that same logic to chan_iax.
(closes issue #14837)
Reported by: barthpbx
Patches:
iax2-rtsavesysname.patch uploaded by barthpbx (license 744)
rt_iax.diff uploaded by dvossel (license 671)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@201534 f38db490-d61c-443f-a65b-d21fe96a405b
also added support for skip category request feature of openr2 and updated chan_dahdi.conf.sample
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@200477 f38db490-d61c-443f-a65b-d21fe96a405b
This code was there because of the AgentCallbackLogin() application.
->loginchan[] member was only used by AgentCallbackLogin().
Agent where dumped to astdb if they where logged in using AgentCallbacklogin()
so they are not being dumper anymore.
Review: https://reviewboard.asterisk.org/r/267/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@198217 f38db490-d61c-443f-a65b-d21fe96a405b
This commit add Calendaring support to Asterisk for iCalendar, CalDAV, and MS
Exchange calendars. Exchange support has only been tested on Exchange Server 2k3
and does not support forms-based authentication at this time (patches *very*
welcome). Exchange support is also currently missing the ability to return a
list of a meting's attendees (again, patches are very, very welcome).
Features include:
Querying a calendar for events over a specific time range
Checking a calendar's busy status via the dialplan
Writing calendar events via the dialplan (CalDAV and Exchange only)
Handling calendar event notifications through the dialplan
(closes issue #14771)
Tested by: lmadsen, twilson, Shivaprakash
Review: https://reviewboard.asterisk.org/r/58
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@197738 f38db490-d61c-443f-a65b-d21fe96a405b
Let's try that again, this time removing trailing whitespace and not leading
whitespace. I can't believe no one noticed.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@197535 f38db490-d61c-443f-a65b-d21fe96a405b
In sip.conf the transport option allows for the configuration of what transport types (udp, tcp, and tls) a peer will accept, but only the first type listed was used for outbound connections. This patch changes this. Now the default transport type is only used until the peer registers. When registration takes place the transport type is parsed out of the Contact header. If the Contact header's transport type is equal to one that the peer supports, the peer's default transport type for outbound connections is set to match the Contact header's type. If the Contact header's transport type is not present, then the peer's default transport type is set to match the one the peer registered with. When a peer unregisters or the registration expires, the default transport type for that peer is reset.
(closes issue #12282)
Reported by: rjain
Patches:
reg_patch_1.diff uploaded by dvossel (license 671)
Tested by: dvossel
(closes issue #14727)
Reported by: pj
Patches:
reg_patch_3.diff uploaded by dvossel (license 671)
Tested by: pj, dvossel
Review: https://reviewboard.asterisk.org/r/249/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@196416 f38db490-d61c-443f-a65b-d21fe96a405b
SIP purists may want to look the other way...
When COLP/CONP support for SIP was committed, there was a condition under
which Asterisk may transmit a SIP UPDATE in order to communicate the change
in connected line information. The issue here is that while we could send a
SIP UPDATE message, we were not prepared to receive such an UPDATE and would
always responde with a 501 when we received an UPDATE.
The situation was a bit rough. We really want to be able to receive UPDATEs
having to do with connected line changes, but the amount of effort involved
in properly supporting RFC 3311 was staggering. This commit represents a
compromise.
First, it was decided that it is important to only send a SIP UPDATE to
an endpoint that is able to handle one. So, now we have added parsing of
the Allow header into SIP. We store the allowed methods on SIP peers so
that when we communicate with them, we already will know what we can and
cannot send to them. We will parse the peer's allowed methods when he registers
with us. If the peer is not the type to register with us, but the qualify option
is enabled, then we will use the response to the OPTIONS request we send
the peer to determine the peer's allowed methods. When the peer's registration
expires, or when qualify deems the peer to be unreachable, we clear the allowed
methods from the peer.
For an actual call, we will copy the peer's allowed methods to the sip_pvt
representing the call leg. If we are communicating with an endpoint which is
not a peer, then we will just parse the Allow header from the first message
we receive during the call and store the information in the sip_pvt.
If, during communication with a peer, we receive a 501 response, then we will
make sure to save the fact that we cannot use that method when communicating
with that peer.
Now, with all that infrastructure in place, the only actual place we use this
information currently is when attempting to send a connected line change using
an UPDATE request. If we cannot send the change immediately using an UPDATE,
we will set the SIP_NEEDREINVITE flag so that we can send a REINVITE as soon
as it is allowed.
The second part of the changes here is for Asterisk to accept UPDATE requests
that have connected line changes. Since we are not fully supporting RFC 3311,
Asterisk will NOT place the UPDATE method in Allow headers it sends. Instead,
if you are communicating with what you know to be another Asterisk box, you may
set the rpid_update parameter in sip.conf so that we will send UPDATEs to that
Asterisk box. When we send a connected line update, we set a custom header
called "X-Asterisk-rpid-update."
On the receiving end, if Asterisk receives an UPDATE that does not have the
"X-Asterisk-rpid-update" header present, then Asterisk will respond with a 501
since media-changing UPDATEs are not supported. We should never get such
UPDATEs, since as was stated earlier, Asterisk does not put UPDATE in its Allow
header. If the custom header is present in the received UPDATE, though, then we
will check the incoming request for connected line updates and queue the update
on the channel where the change occurred.
ABE-1840
ABE-1822
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@195589 f38db490-d61c-443f-a65b-d21fe96a405b
Up to now, cdr_custom would only accept a single filename/format from
cdr_custom.conf. This change allows you to specify multiple filename
& format directives.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@195165 f38db490-d61c-443f-a65b-d21fe96a405b
Select what to do with outgoing COLP information on this port.
0 - Send out COLP information unaltered. (default)
1 - Force COLP to restricted on all outgoing COLP information.
2 - Do not send COLP information.
outgoing_colp=0
Also fixed sending the EctInform message so it always has the
required redirectionNumber parameter when the status is active.
JIRA ABE-1853
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@194479 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r193193 | kpfleming | 2009-05-08 09:03:28 -0500 (Fri, 08 May 2009) | 7 lines
Make absolute paths for logger channels work properly
(Note: This is not a new feature, it was previously undocumented and broken.)
The Asterisk logger has a feature to support absolute pathnames for logger channels, but the code implementing the feature was broken. This has been fixed, and the absolute path feature is now documented in the sample logger.conf.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@193194 f38db490-d61c-443f-a65b-d21fe96a405b
This configuration file was changed to ensure that only one console channel driver
(chan_oss) is loaded by default, but the change would only work if chan_console
was not built. Now it will work as expected; if chan_alsa or chan_console are built
and installed, they will not be loaded unless explicity requested.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@191955 f38db490-d61c-443f-a65b-d21fe96a405b
In discussions today at the Europe Asterisk Developer Meet-Up, we determined that
the event_log was used in only 9 places in the entire tree, and really was not needed
at all. The users have been converted to use LOG_NOTICE, or the messages have been
removed since other messages were already in place that provided the same information.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@191785 f38db490-d61c-443f-a65b-d21fe96a405b
chan_sip allows for outbound TLS connections, but does not allow the user to specify what protocol to use (default was SSLv2, and still is if this new option is not specified). This patch lets the user pick the SSL/TLS client method for outbound connections in sip.
(closes issue #14770)
Reported by: TheOldSaint
(closes issue #14768)
Reported by: TheOldSaint
Review: http://reviewboard.digium.com/r/240/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@191177 f38db490-d61c-443f-a65b-d21fe96a405b
ast_tls_read_conf() is a new api call for handling SSL/TLS options across all conf files. Before this change, SSL/TLS options were not consistent. http.conf and manager.conf required the 'ssl' prefix while sip.conf used options with the 'tls' prefix. While the options had different names in different conf files, they all did the exact same thing. Now, instead of mixing 'ssl' or 'tls' prefixes to do the same thing depending on what conf file you're in, all SSL/TLS options use the 'tls' prefix. For example. 'sslenable' in http.conf and manager.conf is now 'tlsenable' which matches what already existed in sip.conf. Since this has the potential to break backwards compatibility, previous options containing the 'ssl' prefix still work, but they are no longer documented in the sample.conf files. The change is noted in the CHANGES file though.
Review: http://reviewboard.digium.com/r/237/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@191028 f38db490-d61c-443f-a65b-d21fe96a405b
The option to choose which connected line header to
use is not 'rpid_header' but 'sendrpid'
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@190577 f38db490-d61c-443f-a65b-d21fe96a405b
Adds option to specify a private key .pem file when configuring TLS or SSL in AMI, HTTP, and SIP. Before this, the certificate file was used for both the public and private key. It is possible for this file to hold both, but most configurations allow for a separate private key file to be specified. Clarified in .conf files how these options are to be used. The current conf files do not explain how the private key is handled at all, so without knowledge of Asterisk's TLS implementation, it would be hard to know for sure what was going on or how to set it up.
Review: http://reviewboard.digium.com/r/234/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@190545 f38db490-d61c-443f-a65b-d21fe96a405b
This is the companion commit to libpri r732. Service messages are now supported
for switch types 4ess/5ess. A new option service_message_support has been added
to chan_dahdi.conf and is noted in the sample config file. The service message
support is turned off by default. The current implementation relies on AstDB
to keep track of channel state, which allows the statuses to be preserved
across Asterisk restarts. Below is a description of the storage format.
The state and reason for the service state are in the form <state>:<reason>,
where:
<state> ::= { 'O' } // 'O' – Out Of Service
<reason> ::= { '0' | '1' | '2' | '3' }, where:
'0' – No reason (backwards compatibility)
'1' – NEAR END
'2' – FAR END
'3' – both NEAR and FAR END
The new CLI commands to handle channel service state are:
pri service disable channel <chan>
pri service enable channel <chan>
Many people contributed to the development of this functionality. Because I
entered at the very end I do not know the exact history. Special thanks to
all who moved the bug forward one way or another:
cmaj, PCadach, markster, mattf, drmac, MikeJ, serge-v, murf, kanelbullar, Seb7,
tilghman, lmadsen, and especially dhubbard (he answered lots of my questions
and did a large portion of the work)
(closes issue #3450)
Reported by: cmaj
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@188342 f38db490-d61c-443f-a65b-d21fe96a405b
This allows for you to change the From header for outgoing MWI
NOTIFY requests. Prior to this, the best you could do was to
set a callerid in the general section of sip.conf. The problem
was that this was used for all outbound requests, not just
MWI NOTIFY requests.
AST-201
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187560 f38db490-d61c-443f-a65b-d21fe96a405b
(inspired by this evening's Toronto Asterisk Users Group meeting and previous dicussions amongst various community members)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187269 f38db490-d61c-443f-a65b-d21fe96a405b
The channel drivers which have been most heavily tested with these enhancements are
chan_sip and chan_misdn. Further work is being done to add Q.SIG support and will be
introduced in a later commit. chan_skinny has code added to it here, but according
to user pj, the support on chan_skinny is not working as of now. This will be fixed in
a later commit.
A special thanks goes out to bugtracker user gareth for getting the ball rolling and
providing the initial support for this work. Without his initial work on this, this would
not have been nearly as painless as it was.
This functionality has been tested by Digium's product quality department, as well as a
customer site running thousands of calls every day. In addition, many many many many bugtracker
users have tested this, too.
(closes issue #8824)
Reported by: gareth
Review: http://reviewboard.digium.com/r/201
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@186525 f38db490-d61c-443f-a65b-d21fe96a405b
This API provides a generic way for multiple RTP stacks to be
integrated into Asterisk. Right now there is only one present, res_rtp_asterisk,
which is the existing Asterisk RTP stack. Functionality wise this commit
performs the same as previously. API documentation can be viewed in the
rtp_engine.h header file.
Review: http://reviewboard.digium.com/r/209/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@186078 f38db490-d61c-443f-a65b-d21fe96a405b
This commit introduces official support for R2 signaling in chan_dahdi. The
modifications to chan_dahdi, and the supporting library, LibOpenR2, were both
written by Moises Silva.
Many users are using this code, or a variant of it, in Asterisk 1.2, 1.4 and 1.6
in Brazil, México and Argentina. An unknown number of users (but at least 1)
are using it in each of the following countries: Colombia, Nepal, Thailand,
Venezuela, Perú, and probably others.
To use this code, LibOpenR2 must be installed from http://www.libopenr2.org/.
Information about configuration can be found in configs/chan_dahdi.conf.sample.
The code committed is the most up to date version, which was being maintained
in svn/asterisk/team/moy/mfcr2/.
I would also like to include a Thank You to the many others that tested this
code beyond those listed in this commit message. These are the names that I
could find in the mantis issue.
(closes issue #12509)
Reported by: moy
Patches:
chan_zap-mfr2.patch uploaded by moy (license 222)
Tested by: moy, korihor, viniciusfontes, Skarmeth, loloski, asbestoshead, titogarrido, heliocoelhojr, konsultex, ncorrare, ecarruda, rtorresduque, PTorres, ychen
Review: http://reviewboard.digium.com/r/40/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182355 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r180380 | mmichelson | 2009-03-05 12:58:48 -0600 (Thu, 05 Mar 2009) | 25 lines
Fix broken mailbox parsing when searchcontexts option is enabled.
When using the searchcontexts option in voicemail.conf, the code
made the assumption that all mailbox names defined were unique across
all contexts. However, the code did nothing to actually enforce this
assumption, nor did it do anything to alert a user that he may have
created an ambiguity in his voicemail.conf file by defining the same
mailbox name in multiple contexts.
With this change, we now will issue a nice long warning if searchcontexts
is on and we encounter the same mailbox name in multiple contexts and ignore
any duplicates after the first box. Whether searchcontexts is enabled or not,
if we come across a duplicate mailbox in the same context, then we will issue
a warning and ignore the duplicated mailbox. I have also added a small note
to voicemail.conf.sample in the explanation for searchcontexts explaining
that you cannot define the same mailbox in multiple contexts if you have
enabled the option.
(closes issue #14599)
Reported by: lmadsen
Patches:
14599.patch uploaded by mmichelson (license 60) (with slight modification)
Tested by: lmadsen
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@180383 f38db490-d61c-443f-a65b-d21fe96a405b
When the subscription context for a call pickup subscription differs
from the context of the call pickup target, there's not an easy way
to divine what context should be used for the pickup. The way to work
around this is to use PICKUPMARK as the context for the pickup.
This has been documented in the sip.conf.sample file
(ABE-1708)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@180155 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r180006 | mmichelson | 2009-03-03 16:48:18 -0600 (Tue, 03 Mar 2009) | 17 lines
Clarify some documentation of queues.conf.sample
It had always been possible to explicitly specify a "blank"
value for a sound file in queues.conf and have no sound played
back. The problem with this is that it would result in some ugly
CLI warnings from file.c.
This commit introduces a check when playing a file in app_queue
to see if the name of the file is zero-length and return early if
that is the case. Also, the ability to specify the blank sound
files in queues.conf is now mentioned more clearly in queues.conf.sample
(closes issue #14227)
Reported by: caspy
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@180007 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
In this case, it's just a matter of reducing the default timeouts from 2000
to 1000 msec, as the max def feature digit timeout is no longer halved.
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r178956 | murf | 2009-02-26 14:27:32 -0700 (Thu, 26 Feb 2009) | 18 lines
This change moves the default feature digit timeout to 1000 ms from the previous default of 500.
As per bug 14515, a dev discussion arrived at a "mediated concensus"
of a default feature digit timeout of 1.0 sec. Some voted for 1300;
ctooley thought 1500 for distracted phone users in phone booths;
kpfleming put his foot down at 1.0 sec.
Users who found the previous default max delay of 250 msec perfect,
are welcome to override the new default. Notice that I said that
250 msec was the default; wait a minute, you might say, the config
file said it was 500 msec!; well, because of the bug fix for 14515,
we found that 500 msec was actually enforcing a max of 250. The bug
fix would restore 500 msec, but we felt even that was a bit tight
for most users... 2000 msec was pushed earlier by mmichelson, so
that reduces to 1000 msec after the bug fix. Enjoy!
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@178986 f38db490-d61c-443f-a65b-d21fe96a405b
This patch includes a number of changes to the indications API. The primary
motivation for this work was to improve stability. The object management
in this API was significantly flawed, and a number of trivial situations could
cause crashes.
The changes included are:
1) Remove the module res_indications. This included the critical functionality
that actually loaded the indications configuration. I have seen many people
have Asterisk problems because they accidentally did not have an
indications.conf present and loaded. Now, this code is in the core,
and Asterisk will fail to start without indications configuration.
There was one part of res_indications, the dialplan applications, which did
belong in a module, and have been moved to a new module, app_playtones.
2) Object management has been significantly changed. Tone zones are now
managed using astobj2, and it is no longer possible to crash Asterisk by
issuing a reload that destroys tone zones while they are in use.
3) The API documentation has been filled out.
4) The API has been updated to follow our naming conventions.
5) Various bits of code throughout the tree have been updated to account
for the API update.
6) Configuration parsing has been mostly re-written.
7) "Code cleanup"
The code is from svn/asterisk/team/russell/indications/.
Review: http://reviewboard.digium.com/r/149/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@176627 f38db490-d61c-443f-a65b-d21fe96a405b
Turns key rotation back on by default. Added bit into encryption IE to indicate whether or not key rotation is supported or not. If it is not supported then it is not enabled, which insures backwards compatibility. This eliminates the need for the keyrotate option in iax.conf, so it has been removed.
Review: http://reviewboard.digium.com/r/159/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@175597 f38db490-d61c-443f-a65b-d21fe96a405b
When the 'faxdetect' configuration option is used, one may also want to use
the 'faxbuffers' configuration option in chan_dahdi.conf. This option will
dynamically use the configured 'faxbuffers' buffer policy on a channel for
the life of the call following the detection of fax tones. The faxbuffers
buffer policy will be reverted during call teardown.
An example use of 'faxbuffers' is below. This example would switch to using
6 buffers with a full buffer policy.
faxbuffers=>6,full
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@175411 f38db490-d61c-443f-a65b-d21fe96a405b
This patch adds forceencryption=yes as an iax.conf option. When force encryption is enabled, no unencrypted connections are allowed. This insures all connections are encrypted. This is a new feature, so CHANGES and iax.conf.sample are updated as well.
(closes issue #13285)
Reported by: sgofferj
Tested by: russell
Review: http://reviewboard.digium.com/r/150/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@175344 f38db490-d61c-443f-a65b-d21fe96a405b
This is very similar to the DAHDI immediate=yes option. When the phone is picked up, instead of giving a dialtone it connects directly to the "s" extension. Changes where implemented in chan_iax2.c to directly connect to the "s" extension in the appropriate context when this option is enabled. Examples explaining its use are added to iax2.conf.sample. CHANGES has been updated as well.
(closes issue #14266)
Reported by: jcovert
Patches:
chan_iax2.c.patch-trunk uploaded by jcovert (license 551)
iax.conf.sample.patch uploaded by jcovert (license 551)
Tested by: jcovert, dvossel
Review: http://reviewboard.digium.com/r/143/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@174046 f38db490-d61c-443f-a65b-d21fe96a405b
In trunk, the only necessary change pointed out was that the call
to ChanIsAvail uses an option that has been removed.
For the 1.6.1 branch, however, it appears that the sample file is
badly in need of updating since there are |'s used all over the place
there. My tentative plan is just to copy trunk's sample config file
to those branches since the info there is most up-to-date and should
be correct for use in 1.6.1
Thanks to macli in #asterisk-dev for bringing this up
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@173776 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r172517 | twilson | 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009) | 37 lines
Fix feature inheritance with builtin features
When using builtin features like parking and transfers, the AST_FEATURE_* flags
would not be set correctly for all instances when either performing a builtin
attended transfer, or parking a call and getting the timeout callback. Also,
there was no way on a per-call basis to specify what features someone should
have on picking up a parked call (since that doesn't involve the Dial() command).
There was a global option for setting whether or not all users who pickup a
parked call should have AST_FEATURE_REDIRECT set, but nothing for DISCONNECT,
AUTOMON, or PARKCALL.
This patch:
1) adds the BRIDGE_FEATURES dialplan variable which can be set either in the
dialplan or with setvar in channels that support it. This variable can be set
to any combination of 't', 'k', 'w', and 'h' (case insensitive matching of the
equivalent dial options), to set what features should be activated on this
channel. The patch moves the setting of the features datastores into the
bridging code instead of app_dial to help facilitate this.
2) adds global options parkedcallparking, parkedcallhangup, and
parkedcallrecording to be similar to the parkedcalltransfers option for
globally setting features.
3) has builtin_atxfer call builtin_parkcall if being transfered to the parking
extension since tracking everything through multiple masquerades, etc. is
difficult and error-prone
4) attempts to fix all cases of return calls from parking and completed builtin
transfers not having the correct permissions
(closes issue #14274)
Reported by: aragon
Patches:
fix_feature_inheritence.diff.txt uploaded by otherwiseguy (license 396)
Tested by: aragon, otherwiseguy
Review http://reviewboard.digium.com/r/138/
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@172580 f38db490-d61c-443f-a65b-d21fe96a405b
* Added doxygen comments to the major dahdi structures.
* Fixed PRI and SS7 using an incorrect string value if the extension
delimiter is not present in the Dial() function.
* Fixed SS7 not checking if the dialed extension is at least as long
as the stripmsd option.
* Fixed PRI not handling unknown TON/NPI prefix letters correctly.
* Fixed some uninitialized string variables on FXS ports.
configs/chan_dahdi.conf.sample
* Updated some documentation.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@172400 f38db490-d61c-443f-a65b-d21fe96a405b
This prevents the situation when MWI messages are added to caller ID spills causing the channel to be hung up
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@169153 f38db490-d61c-443f-a65b-d21fe96a405b
With this commit, a register => line in sip.conf may contain a port number in the
"user" section of the line. Please see CHANGES and sip.conf.sample for more
details regarding this.
(closes issue #14198)
Reported by: Nick_Lewis
Patches:
chan_sip.c-domainport2.patch uploaded by Nick (license 657)
Tested by: Nick_Lewis
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@168575 f38db490-d61c-443f-a65b-d21fe96a405b
Update the queues.conf.sample documentation to mention that you need to preload chan_local.so as well if you plan on using Local channels for queue members, and you're preloading pbx_config.so.
(closes issue #14179)
Reported by: CrashHD
Tested by: CrashHD
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@167477 f38db490-d61c-443f-a65b-d21fe96a405b
enabled (either globally or for a specific peer), chan_sip will treat any SDP
data it receives as new data and update the media stream accordingly. By
default, Asterisk will only modify the media stream if the SDP session version
received is different from the current SDP session version. This option is
required to interoperate with devices that have non-standard SDP session
version implementations (observed by toc on the bug tracker with Microsoft OCS
which always uses 0 as the session version).
http://reviewboard.digium.com/r/94/
(closes issue #13958)
Reported by: toc
Tested by: toc
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@165180 f38db490-d61c-443f-a65b-d21fe96a405b
Terry Wilson created the original patch for this functionality, which I slightly modified and added
the faxdetect=yes|no configuration option. This patch is only for T38 fax detection and does not
do anything for G711 over SIP fax detection. By default, this option is disabled.
Reviewboard: http://reviewboard.digium.com/r/69/
This functionality is for issue AST-140.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@161115 f38db490-d61c-443f-a65b-d21fe96a405b
Based on cli_permissions.conf configuration file, we are able to permit or deny
cli commands based on some patterns and the local user and group running rasterisk.
(Sorry if I missed some of the testers).
Reviewboard: http://reviewboard.digium.com/r/11/
(closes issue #11123)
Reported by: eliel
Tested by: eliel, IgorG, Laureano, otherwiseguy, mvanbaak
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@160062 f38db490-d61c-443f-a65b-d21fe96a405b
is only found if it dialed the extension that was subscribed to. You can now
specify 'ignore-context' for the 'notifycid' option in sip.conf which will, as
it's value implies, ignore the current context of the caller when doing the
lookup.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@158756 f38db490-d61c-443f-a65b-d21fe96a405b
- Add CLI aliases module to asterisk.
- Remove all deprecated CLI commands from the code
Initial work done by file.
Junk-Y and lmadsen did a lot of work and testing to
get the list of deprecated commands into the configuration file.
Deprecated CLI commands are now handled by this new module,
see cli_aliases.conf for more info about that.
ok russellb@ via reviewboard
(closes issue #13735)
Reported by: mvanbaak
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@156120 f38db490-d61c-443f-a65b-d21fe96a405b
lying about what's possible. The tz cannot be set in a
context like this. It can only be set in the general
section or per-mailbox.
Thanks to sasargen on #asterisk-dev for pointing this out
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@155360 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r155011 | mmichelson | 2008-11-06 13:45:52 -0600 (Thu, 06 Nov 2008) | 8 lines
The documentation listed the ability to set 'maxmsg' per
context. The truth is that you can only set this in the general section
or per mailbox. Thus I am updating the sample config file to be more
accurate.
Thanks to sasargen on IRC for bringing up this issue.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@155012 f38db490-d61c-443f-a65b-d21fe96a405b
the calling party when subscribed to the state of an extension that is ringing.
This has some limitations which are documented in sip.conf.sample.
(closes issue #13827)
Reported by: seanbright
Patches:
issue13827.patch uploaded by seanbright (license 71)
Reviewed by: russellb
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@154187 f38db490-d61c-443f-a65b-d21fe96a405b
remotesecret => our password for a remote service
secret => our authentication when someone calls us
Secret => still has both functions if remotesecret is not used.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@153904 f38db490-d61c-443f-a65b-d21fe96a405b
the low default value of featuredigittimeout, I decided it
was high time to change it. I have changed the default to
2000 ms based on a suggestion from Leif Madsen.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@152807 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r152538 | murf | 2008-10-28 23:19:04 -0600 (Tue, 28 Oct 2008) | 14 lines
A little documentation cross-ref between features and
dial and queue... I wasted some time (stupidly) trying
to get the one-touch parking stuff working, because it
didn't occur to me that I had to also have the corresponding
options in the dial command! Duh! (In all this time, I never
set this up before!)
So, to keep some poor fool from suffering the same fate,
I made the features.conf.sample file mention the corresponding
opts in dial/queue; and the docs for dial/app specifically
mention the corresponding decls in the feature.conf file.
I hope this doesn't spoil some vast, eternal plan...
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@152605 f38db490-d61c-443f-a65b-d21fe96a405b
and "leavewhenempty" options are configured in queues.conf.
Instead of using vague terms like "yes," "no," "loose," and
"strict," we now accept a comma-separated list of values
to determine when to consider a member available.
Extended details can be found in the queues.conf.sample
file. Note also that the above four referenced values are
still accepted for backwards-compatibility, but are mapped
internally to the new method of representing the option.
AST-105
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@146640 f38db490-d61c-443f-a65b-d21fe96a405b
reorganized sip_notify.conf.sample a bit as well. Tested snom
reboot on snom 360 and verified snom-check-cfg worked as well.
(closes issue #13601)
Reported by: mjc
Tested by: seanbright
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@146312 f38db490-d61c-443f-a65b-d21fe96a405b
It was pretty sparsely documented.
This update fleshes out the pbx_lua module, to
add the switch statements to the extensions in the
extensions.lua file, as well as removing them when
the module is unloaded.
Many thanks to Matt Nicholson for his fine
contribution!
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@144523 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r142865 | tilghman | 2008-09-12 15:37:18 -0500 (Fri, 12 Sep 2008) | 11 lines
Create rules for disallowing contacts at certain addresses, which may
improve the security of various installations. As this does not change
any default behavior, it is not classified as a direct security fix for
anything within Asterisk, but may help PBX admins better secure their
SIP servers.
(closes issue #11776)
Reported by: ibc
Patches:
20080829__bug11776.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76, blitzrage
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@142866 f38db490-d61c-443f-a65b-d21fe96a405b
Reported by: erousseau
This was a reasonable enhancement request, which was
easy to implement. Since it's an enhancement, it
could only be applied to trunk.
Basically, for accounting where "initiated" seconds
are billed for, if the microseconds field on the end
time is greater than the microseconds field for the
answer time, add one second to the billsec field.
The implementation was requested by erousseau, and
I've implemented it as requested. I've updated the
CHANGES, the cdr.conf.sample, and the .h files
accordingly, to accept and set a flag for the
corresponding new option. cdr.c adds in the extra
second based on the usec fields if the option is
set. Tested, seems to be working fine.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@140057 f38db490-d61c-443f-a65b-d21fe96a405b
* Made bearer2str() use allowed_bearers_array[]
* Made use the causes.h defines instead of hardcoded numbers.
* Made use Asterisk presentation indicator values if either of the
mISDN presentation or screen options are negative.
* Updated the misdn_set_opt application option descriptions.
* Renamed the awkward Caller ID presentation misdn_set_opt
application option value not_screened to restricted.
Deprecated the not_screened option value.
channels/misdn/isdn_lib.c
* Made use the causes.h defines instead of hardcoded numbers.
* Fixed some spelling errors and typos.
* Added all defined facility code strings to fac2str().
channels/misdn/isdn_lib.h
* Added doxygen comments to struct misdn_bchannel.
channels/misdn/isdn_lib_intern.h
* Added doxygen comments to struct misdn_stack.
channels/misdn_config.c
configs/misdn.conf.sample
* Updated the mISDN presentation and screen parameter descriptions.
doc/tex/misdn.tex
* Updated the misdn_set_opt application option descriptions.
* Fixed some spelling errors and typos.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@138738 f38db490-d61c-443f-a65b-d21fe96a405b
the port configuration option from cdr_tds.conf. So go ahead and
remove it from the sample config.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@138442 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r137731 | russell | 2008-08-14 09:05:23 -0500 (Thu, 14 Aug 2008) | 4 lines
Comments in this config file were aligned only if your tab size was set to 8.
So, convert tabs to spaces so that things should be aligned regardless of what
tab size you use in your editor.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@137732 f38db490-d61c-443f-a65b-d21fe96a405b
This set of changes enhances IAX2 encryption support by adding key rotation
to provide enhanced security. The key used for encryption is rotated right
after the call gets set up, and then again every few minutes. This was
discussed at the last AstriDevCon. For interoperability with older versions
of Asterisk, there is an option that disables key rotation.
(closes issue #13018)
Reported by: bbryant
Patches:
07072008__iax2_key_rotation.diff uploaded by bbryant (license 36)
Tested by: russell, bbryant
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135158 f38db490-d61c-443f-a65b-d21fe96a405b
such as "Work" and "Family" would be subfolders of the
INBOX. This is an invalid assumption to make, but it could
be desirable to set up folders in this manner, so a new
option for voicemail.conf, "imapparentfolder" has been
added to allow for this.
(closes issue #13142)
Reported by: jaroth
Patches:
parentfolder.patch uploaded by jaroth (license 50)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135067 f38db490-d61c-443f-a65b-d21fe96a405b
driver into a common place for multiple channel drivers.
(closes issue #13152)
Reported by: caio1982
Patches:
atxfer_complete_sound3.diff uploaded by caio1982 (license 22)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@134401 f38db490-d61c-443f-a65b-d21fe96a405b
fail to setup video RTP if the two endpoints will not support it. This assists
with call files and certain transfers to ensure that if two video phones are
ever connected, they will always share a video feed.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@130951 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r130039 | kpfleming | 2008-07-11 10:41:56 -0500 (Fri, 11 Jul 2008) | 4 lines
add support for a configuration parameter for 'inband audio during RELEASE', which is currently mandatory in libpri-1.4.4 but will become configurable in libpri-1.4.5 later today
(related to issue #13042)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@130040 f38db490-d61c-443f-a65b-d21fe96a405b
Note: I don't think we can start properly without UDP port open, that needs to be tested.
- Removing "bindport" from configuration example, not needed to mention this any more
I suggest we deprecate "bindaddr" and "bindport" in trunk (for 1.6.1)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128525 f38db490-d61c-443f-a65b-d21fe96a405b
- Adding IP address for TCP and/or TLS too if auto-domain is enabled and
binding to a different IP address
- Fixing documentation in sip.conf.sample
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128524 f38db490-d61c-443f-a65b-d21fe96a405b
acceptdtmf and enddtmf. These allow for the DTMF pressed to be configurable
instead of being hardcoded to '#' and '*'.
(AST-86)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@127558 f38db490-d61c-443f-a65b-d21fe96a405b
for new dialogs, and the rest listed will be acceptable ways for that peer to contact us. This fixes a minor bug where, because SIP TCP/UDP run on
the same port, could cause a TCP peer to be saved in the ast_db. There will also be warnings when a transport is changed for an unexpected reason.
(issue #12799)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@125891 f38db490-d61c-443f-a65b-d21fe96a405b
This commit merges in the rest of the code needed to support distributed device
state. There are two main parts to this commit.
Core changes:
- The device state handling in the core has been updated to understand device
state across a cluster of Asterisk servers. Every time the state of a device
changes, it looks at all of the device states on each node, and determines the
aggregate device state. That resulting device state is what is provided to
modules in Asterisk that take actions based on the state of a device.
New module, res_ais:
- A module has been written to facilitate the communication of events between
nodes in a cluster of Asterisk servers. This module uses the SAForum AIS
(Service Availability Forum Application Interface Specification) CLM and EVT
services (Cluster Management and Event) to handle this task. This module
currently supports sharing Voicemail MWI (Message Waiting Indication) and
device state events between servers. It has been tested with openais, though
other implementations of the spec do exist.
For more information on testing distributed device state, see the following doc:
- doc/distributed_devstate.txt
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@121559 f38db490-d61c-443f-a65b-d21fe96a405b
tilghman