Joshua Colp
4897378ff1
Merged revisions 72331 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r72331 | file | 2007-06-27 18:58:02 -0400 (Wed, 27 Jun 2007) | 2 lines
Make payload IDs for iLBC/Speex match to our list. Since these are dynamic payloads the other side shouldn't care. (issue #9426 reported by irroot)
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2007-06-27 22:58:53 +00:00
Jason Parker
8bf745194f
Merged revisions 72125 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r72125 | qwell | 2007-06-27 12:10:32 -0500 (Wed, 27 Jun 2007) | 4 lines
Don't modify a variable that we don't want modified. Make a copy of it instead.
Issue 10029, patch by phsultan with slight modifications by me (to remove needless casts).
Note: chan_jingle in trunk does not appear to have the same bug.
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2007-06-27 17:14:31 +00:00
Russell Bryant
3957ce9215
Merged revisions 70084 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r70084 | russell | 2007-06-19 14:13:45 -0500 (Tue, 19 Jun 2007) | 3 lines
Only attempt to queue a hangup on the owner channel if it actually exists.
(issue #9795 , patch from zandbelt)
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2007-06-19 19:15:03 +00:00
Russell Bryant
055d82cbce
Add a massive set of changes for converting to use the ast_debug() macro.
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(issue #9957 , patches from mvanbaak, caio1982, critch, and dimas)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69327 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-14 19:39:12 +00:00
Tilghman Lesher
9d05ff8ed5
Issue 9869 - replace malloc and memset with ast_calloc, and other coding guidelines changes
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2007-06-06 21:20:11 +00:00
Tilghman Lesher
ce9ec91897
ast_calloc janitor (Inspired by issue 9860)
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2007-06-03 06:10:27 +00:00
Kevin P. Fleming
c74518e3ff
Merged revisions 66157 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r66157 | kpfleming | 2007-05-25 10:28:46 -0400 (Fri, 25 May 2007) | 3 lines
handle the GNUTLS library properly in the configure script and build system
don't build in OSP support unless we have found and are allowed to use SSL support
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2007-05-25 14:37:55 +00:00
Kevin P. Fleming
287bcc6127
Merged revisions 65965-65967 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r65965 | kpfleming | 2007-05-24 14:24:55 -0400 (Thu, 24 May 2007) | 2 lines
don't use uninitialized variables
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r65966 | kpfleming | 2007-05-24 14:25:21 -0400 (Thu, 24 May 2007) | 2 lines
don't reference GnuTLS headers and functions unless the configure script found it
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r65967 | kpfleming | 2007-05-24 14:28:48 -0400 (Thu, 24 May 2007) | 2 lines
oops, use #ifdef instead of #if
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2007-05-24 19:05:42 +00:00
Olle Johansson
b4f7d35240
Merged revisions 65901 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r65901 | oej | 2007-05-24 17:26:10 +0200 (Thu, 24 May 2007) | 2 lines
Issue 7672 - fix by zandbelt - Asterisk core dump since the GnuTLS interface did not support multithreading correctly.
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2007-05-24 15:29:10 +00:00
Olle Johansson
bd2ae8b587
Merged revisions 65892 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r65892 | oej | 2007-05-24 17:20:54 +0200 (Thu, 24 May 2007) | 2 lines
Issue 8193 - NAT issues with gtalk/STUN. Patch by phsultan. Thanks!
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2007-05-24 15:23:04 +00:00
Olle Johansson
45bb9430b9
Merged revisions 65857 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r65857 | oej | 2007-05-24 17:05:10 +0200 (Thu, 24 May 2007) | 2 lines
Issue 7686, fix by phsultan, NAT issues when calling from gtalk to SIP over nat.
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2007-05-24 15:21:39 +00:00
Olle Johansson
d3c7be548d
Merged revisions 65841 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r65841 | oej | 2007-05-24 16:48:55 +0200 (Thu, 24 May 2007) | 2 lines
Issue #8536 - Caller ID not set in CDR for jingle
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2007-05-24 14:52:01 +00:00
Steve Murphy
ecaf781933
Merged revisions 60989 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r60989 | murf | 2007-04-09 12:32:07 -0600 (Mon, 09 Apr 2007) | 1 line
This is a big improvement over the current CDR fixes. It may still need refinement, but this won't have as many folks bothered.
This also adds the mods from 1.4/r.61136;
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2007-04-10 05:41:34 +00:00
Russell Bryant
b2ddaaf033
Add support for RTP packetization in chan_jingle and chan_gtalk.
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(issue #9416 , phsultan)
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2007-04-03 22:33:03 +00:00
Jason Parker
28a6129af8
Merged revisions 55954 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r55954 | qwell | 2007-02-21 14:27:08 -0600 (Wed, 21 Feb 2007) | 4 lines
Fix locking issue, and accept "transport-accept" as a valid accept message.
This should solve issues 8970 and 8503.
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2007-02-21 20:30:54 +00:00
Jason Parker
8f28800765
Merged revisions 55799 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r55799 | qwell | 2007-02-20 20:01:36 -0600 (Tue, 20 Feb 2007) | 4 lines
Fix segfault when buddy couldn't be found.
Issue 7764, patch by sailer
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2007-02-21 02:04:10 +00:00
Jason Parker
ae47fc4541
Merged revisions 55555 via svnmerge from
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r55555 | qwell | 2007-02-20 10:53:45 -0600 (Tue, 20 Feb 2007) | 4 lines
No need to cast nor free with strdupa (thanks file)
55555!
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2007-02-20 16:56:58 +00:00
Olle Johansson
ba32ee49d0
Adding Realtime Text support (T.140) to Asterisk
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T.140/RFC 2793 is a live communication channel, originally
created for IP based text phones for hearing impaired.
Feels very much like the old Unix talk application.
This code is developed and disclaimed by John Martin of Aupix, UK.
Tested for interoperability by myself and Omnitor in Sweden,
the company that wrote most of the specifications.
A big thank you to everyone involved in this.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54838 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-16 13:35:44 +00:00
Kevin P. Fleming
e758ef09b3
Merged revisions 53779-53781 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53779 | kpfleming | 2007-02-09 17:51:29 -0600 (Fri, 09 Feb 2007) | 2 lines
fix awk scripts to work when both MODULEINFO and MAKEOPTS are present in a source file
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r53780 | kpfleming | 2007-02-09 17:51:41 -0600 (Fri, 09 Feb 2007) | 2 lines
add some inter-module dependencies
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r53781 | kpfleming | 2007-02-09 17:52:44 -0600 (Fri, 09 Feb 2007) | 2 lines
another dependency
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2007-02-09 23:53:51 +00:00
Joshua Colp
ee3ab150f6
Merged revisions 51788 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r51788 | file | 2007-01-23 17:46:31 -0500 (Tue, 23 Jan 2007) | 2 lines
Update channel drivers to use module referencing so that unloading them while in use will not result in crashes. (issue #8897 reported by junky)
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2007-01-23 22:59:55 +00:00
Russell Bryant
3275357f20
Merged revisions 51328 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r51328 | russell | 2007-01-19 13:08:25 -0600 (Fri, 19 Jan 2007) | 5 lines
Fix VLDTMF support in chan_gtalk. AST_FRAME_DTMF and AST_FRAME_DTMF_END are
actually the same thing. So, a digit would have been interpreted incorrectly
here. Since the channel driver will always have the begin and end callbacks
called for a digit, only support the button-down and button-up messages.
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2007-01-19 19:09:04 +00:00
Russell Bryant
dcca8f345f
Merged revisions 51311 via svnmerge from
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r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines
Merge the changes from the /team/group/vldtmf_fixup branch.
The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged. So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio. However,
since there was no audio coming in, the DTMF_END was never generated. This
caused DTMF based features to no longer work.
To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf). If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.
Channel drivers also now get passed the length of the digit to their digit_end
callback. This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.
(issue #8597 , maybe others...)
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2007-01-19 18:06:03 +00:00
Russell Bryant
ec8591d04c
Constify a bunch of usage strings for CLI commands.
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2006-12-06 07:35:31 +00:00
Joshua Colp
869101028b
Merged revisions 48168 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48168 | file | 2006-11-30 16:18:24 -0500 (Thu, 30 Nov 2006) | 2 lines
Do not do a partial bridge for Google Talk since we need to handle STUN. (issue #8448 reported by phsultan)
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2006-11-30 21:22:01 +00:00
Luigi Rizzo
c8597704ce
fix compilation.
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Overall i think the previous change to ast_channel_alloc()
to close bug 7506 should have been done by defining
an ast_set_callerid_noevent() function that does the
setting without generating the event.
Lot less code duplication, and easier to handle.
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2006-11-08 07:21:45 +00:00
Steve Murphy
908f176cf3
A fair number of changes for the sake of bug 7506
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2006-11-07 21:47:49 +00:00
Luigi Rizzo
39d94767d7
remove useless usecnt stuff
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2006-11-03 12:24:08 +00:00
Matt O'Gorman
09ef9b465b
Merged revisions 46822 via svnmerge from
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https://svn.digium.com/svn/asterisk/branches/1.4
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r46822 | mogorman | 2006-11-01 14:35:41 -0600 (Wed, 01 Nov 2006) | 2 lines
bind address support from bug 8164
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2006-11-01 20:38:05 +00:00
Matt O'Gorman
e20bb6fa69
Merged revisions 44982 via svnmerge from
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r44982 | mogorman | 2006-10-12 15:34:49 -0500 (Thu, 12 Oct 2006) | 2 lines
fix for bug 7764.
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2006-10-12 20:41:37 +00:00
Matt O'Gorman
7294ba3852
Merged revisions 44312 via svnmerge from
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r44312 | mogorman | 2006-10-03 17:35:43 -0500 (Tue, 03 Oct 2006) | 2 lines
fix issue with dialing client without resource.
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2006-10-03 22:36:51 +00:00
Matt O'Gorman
ae8cc3e18b
bug #8076 check option_debug before printing to debug channel.
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patch provided in bugnote, with minor changes.
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2006-10-03 15:53:07 +00:00
Matt O'Gorman
08368f00ab
Merged revisions 43466 via svnmerge from
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r43466 | mogorman | 2006-09-21 18:50:56 -0500 (Thu, 21 Sep 2006) | 2 lines
updates for better compontent support
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2006-09-21 23:55:13 +00:00
Matt O'Gorman
ec4bf7a849
seperate jingle and gtalk so it will be easier to track
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changes in both of the moving specs. Currently chan_gtalk is
compatible with the latest gtalk/libjingle version, and chan_jingle
needs a lot of work.
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2006-09-18 16:36:14 +00:00