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path: root/channels/chan_gtalk.c
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2007-01-23Merged revisions 51788 via svnmerge from Joshua Colp
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
2007-01-19Merged revisions 51328 via svnmerge from Russell Bryant
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
2007-01-19Merged revisions 51311 via svnmerge from Russell Bryant
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
2006-12-06Constify a bunch of usage strings for CLI commands.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48306 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-30Merged revisions 48168 via svnmerge from Joshua Colp
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
2006-11-08fix compilation.Luigi Rizzo
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
2006-11-07A fair number of changes for the sake of bug 7506Steve Murphy
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47290 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-03remove useless usecnt stuffLuigi Rizzo
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47077 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-01Merged revisions 46822 via svnmerge from Matt O'Gorman
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
2006-10-12Merged revisions 44982 via svnmerge from Matt O'Gorman
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
2006-10-03Merged revisions 44312 via svnmerge from Matt O'Gorman
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
2006-10-03bug #8076 check option_debug before printing to debug channel.Matt O'Gorman
patch provided in bugnote, with minor changes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44253 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-09-21Merged revisions 43466 via svnmerge from Matt O'Gorman
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
2006-09-18seperate jingle and gtalk so it will be easier to trackMatt O'Gorman
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
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