asterisk

Commit Graph

Author	SHA1	Message	Date
Kevin P. Fleming	287bcc6127	Merged revisions 65965-65967 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r65965 \| kpfleming \| 2007-05-24 14:24:55 -0400 (Thu, 24 May 2007) \| 2 lines don't use uninitialized variables ........ 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 ........ r65967 \| kpfleming \| 2007-05-24 14:28:48 -0400 (Thu, 24 May 2007) \| 2 lines oops, use #ifdef instead of #if ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65979 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	b4f7d35240	Merged revisions 65901 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65904 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	bd2ae8b587	Merged revisions 65892 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65898 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	45bb9430b9	Merged revisions 65857 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65894 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	d3c7be548d	Merged revisions 65841 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65844 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Steve Murphy	ecaf781933	Merged revisions 60989 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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; ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61152 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	b2ddaaf033	Add support for RTP packetization in chan_jingle and chan_gtalk. (issue #9416, phsultan) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@60011 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Jason Parker	28a6129af8	Merged revisions 55954 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@55955 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Jason Parker	8f28800765	Merged revisions 55799 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@55805 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Jason Parker	ae47fc4541	Merged revisions 55555 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@55556 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	ba32ee49d0	Adding Realtime Text support (T.140) to Asterisk 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	19 years ago
Kevin P. Fleming	e758ef09b3	Merged revisions 53779-53781 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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 ........ r53780 \| kpfleming \| 2007-02-09 17:51:41 -0600 (Fri, 09 Feb 2007) \| 2 lines add some inter-module dependencies ........ r53781 \| kpfleming \| 2007-02-09 17:52:44 -0600 (Fri, 09 Feb 2007) \| 2 lines another dependency ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53782 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	ee3ab150f6	Merged revisions 51788 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51801 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	3275357f20	Merged revisions 51328 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51329 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	dcca8f345f	Merged revisions 51311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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...) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51314 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	ec8591d04c	Constify a bunch of usage strings for CLI commands. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48306 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	869101028b	Merged revisions 48168 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48169 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Luigi Rizzo	c8597704ce	fix compilation. 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. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47306 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Steve Murphy	908f176cf3	A fair number of changes for the sake of bug 7506 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47290 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Luigi Rizzo	39d94767d7	remove useless usecnt stuff git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47077 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Matt O'Gorman	09ef9b465b	Merged revisions 46822 via svnmerge from https://svn.digium.com/svn/asterisk/branches/1.4 ........ r46822 \| mogorman \| 2006-11-01 14:35:41 -0600 (Wed, 01 Nov 2006) \| 2 lines bind address support from bug 8164 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46823 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Matt O'Gorman	e20bb6fa69	Merged revisions 44982 via svnmerge from https://svn.digium.com/svn/asterisk/branches/1.4 ........ r44982 \| mogorman \| 2006-10-12 15:34:49 -0500 (Thu, 12 Oct 2006) \| 2 lines fix for bug 7764. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44983 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Matt O'Gorman	7294ba3852	Merged revisions 44312 via svnmerge from https://svn.digium.com/svn/asterisk/branches/1.4 ........ r44312 \| mogorman \| 2006-10-03 17:35:43 -0500 (Tue, 03 Oct 2006) \| 2 lines fix issue with dialing client without resource. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44313 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Matt O'Gorman	ae8cc3e18b	bug #8076 check option_debug before printing to debug channel. patch provided in bugnote, with minor changes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44253 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Matt O'Gorman	08368f00ab	Merged revisions 43466 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43466 \| mogorman \| 2006-09-21 18:50:56 -0500 (Thu, 21 Sep 2006) \| 2 lines updates for better compontent support ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43467 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Matt O'Gorman	ec4bf7a849	seperate jingle and gtalk so it will be easier to track 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. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43185 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago

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76 Commits (98f42ac6faf429ce0f1c0045c8ba1e6e2ffc60ff)