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path: root/channels/chan_jingle.c
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2007-10-19Convert NEW_CLI to AST_CLI.Jason Parker
Closes issue #11039, as suggested by seanbright. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-16Fix CLI help outputPhilippe Sultan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85787 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-16Added two CLI functions, taken from chan_gtalk :Philippe Sultan
- jingle reload ; - jingle show channels. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85778 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-16Make an audio path under the following call configuration :Philippe Sultan
SIP Phone 1 --- [chan_sip]Asterisk 1[chan_jingle] --- [chan_jingle]Asterisk 2[chan_sip] --- SIP Phone 2 Modifications : - set bridge type to partial ; - process media candidates from the remote peer properly. Now we have Jingle audio, at least between two Asterisk Jingle clients. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85777 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-15Allow RTP structure registrationPhilippe Sultan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85555 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-09Remove redundant includes (patch by snuffy) (Closes issue #10922)Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85140 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-09-25Comply with latest XEP-0166, XEP-0167, XEP-0176.Philippe Sultan
No real Jingle implementation being available, testing was made using two Asterisk servers relaying SIP calls over their Jingle channels: SIP Phone 1 --- [chan_sip]Asterisk 1[chan_jingle] --- [chan_jingle]Asterisk 2[chan_sip] --- SIP Phone 2 Thus, it was possible to test the code in both ways, and make the Jingle channel comply with the latest specifications. No sound available yet. Main modifications include : - modified the 'jingle_candidate' structure and the 'jingle_create_candidates' function according to XEP-0176 ; - modified the 'jingle_action' function in order to properly terminate a Jingle session, in conformance with XEP-0166 ; - modified username format used in STUN requests ; - actually make the bindaddr configuration field useable. Todo : - set audio paths up (no native bridging) ; - make the CLI gtalk functions available to jingle ; - clean up the storage space used in strings. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@83743 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-09-19Replace Google namespace occurrences with Jingle. The former namespacePhilippe Sultan
is handled by chan_gtalk. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@83076 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-09-19Remove namespaces in payload-type tags.Philippe Sultan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@83072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-09-19Transmit proper invitation, thus conforming to XEP-0166 (Jingle generalPhilippe Sultan
specifications), XEP-0167 (Jingle Audio via RTP) and XEP-0176 (Jingle ICE Transport). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@83055 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-09-14Fix DTMF following what has been done in issue #9401. Thanks irroot.Philippe Sultan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82373 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-09-13Modify rule filters to match with the Jingle namespace constantPhilippe Sultan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82320 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-09-13Changed Jingle and Jingle DTMF namespaces.Philippe Sultan
As both specifications are in the Experimental status, the namespaces specified therein shall be of the form "http://www.xmpp.org/extensions/xep-XXXX.html#ns". See the Namespace issuance section in XEP-0053 : http://www.xmpp.org/extensions/xep-0053.html#namespaces git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-09-13Reflect Jingle DTMF specification changesPhilippe Sultan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82312 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-16Don't reload a configuration file if nothing has changed.Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@79747 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-13Merged revisions 79174 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79174 | file | 2007-08-13 11:18:04 -0300 (Mon, 13 Aug 2007) | 4 lines (closes issue #10437) Reported by: haklin Don't set the callerid name and number a second time on a newly created channel. ast_channel_alloc itself already sets it and setting it twice would cause a memory leak. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@79175 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-08Add support for using epoll instead of poll. This should increase ↵Joshua Colp
scalability and is done in such a way that we should be able to add support for other poll() replacements. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@78683 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-27Silly jingle...Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@72358 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-19Merged revisions 70084 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@70088 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-14Add a massive set of changes for converting to use the ast_debug() macro.Russell Bryant
(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
2007-06-03ast_calloc janitor (Inspired by issue 9860)Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66981 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-25more minor fixesKevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66175 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-25Merged revisions 66157 via svnmerge from Kevin P. Fleming
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-07Adding external referenses for doxygenOlle Johansson
See http://www.asterisk.org/doxygen/trunk/extref.html git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63230 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-10updated ast_channel_alloc() call to include the 4 extra args everyone got. ↵Steve Murphy
Not much info there, as the config file evidently does not allow amaflags, or accountcode settings; and the pvt's exten doesn't sound like what we need in the cdr, either. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61221 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-03Add support for RTP packetization in chan_jingle and chan_gtalk.Russell Bryant
(issue #9416, phsultan) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@60011 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-21Merged revisions 55954 via svnmerge from Jason Parker
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
2007-02-21Merged revisions 55799 via svnmerge from Jason Parker
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
2007-02-20Merged revisions 55555 via svnmerge from Jason Parker
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
2007-02-17Update chan_jingle to new definition of set_rtp_peer.Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@55088 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-10add another dependencyRussell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53785 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
2006-11-08fix 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
2006-10-03bug #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
2006-09-18seperate 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
2006-08-31everything that loads a config that needs a config file to runMatt O'Gorman
now reports AST_MODULE_LOAD_DECLINE when loading if config file is not there, also fixed an error in res_config_pgsql where it had a non static function when it should. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41633 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-31Merge in VLDTMF support with Zaptel/Core done by the ever great Darumkilla ↵Joshua Colp
Russell Bryant and the RTP portion done by myself, Muffinlicious Joshua Colp. This has gone through so many discussions/revisions it's not funny but we finally have it! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41507 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-29update to reflect recent rtp changesRussell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-21merge new_loader_completion branch, including (at least):Kevin P. Fleming
- restructured build tree and makefiles to eliminate recursion problems - support for embedded modules - support for static builds - simpler cross-compilation support - simpler module/loader interface (no exported symbols) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@40722 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-16move the calls to ast_jb_configure() to before the PBX thread is started on theRussell Bryant
channel to remove the theoretical race condition that the channel could get bridged before the channel's jitterbuffer gets configured. This was pointed out by PCadach on IRC. Thanks! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39964 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-08some code clean up and catch for a act_hook being calledMatt O'Gorman
without a packet. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39351 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-07Many many code cleanup changes given to me by OejMatt O'Gorman
Thanks, sorry I didn't put this in forever ago. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39229 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-02dtmf support. not everything else, trying to clear out those other bugsMatt O'Gorman
but more to come i guess. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38714 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-21Merge a new implementation of ast_inet_ntoa, our thread safe replacement forRussell Bryant
inet_ntoa, which uses thread specific data (aka thread local storage) instead of stack allocatted buffers to store the result. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-19merge Russell's 'hold_handling' branch, finally implementing music-on-hold ↵Kevin P. Fleming
handling the way it was decided at AstriDevCon Europe 2006 (and the way people really want it to be) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-13allow users of RTP to use G726-32 AAL2 packing even when RFC3551 packing has ↵Kevin P. Fleming
been requested (Sipura/Grandstream ATAs and others will need this) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37501 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-03Blocked revisions 36725 via svnmergeRussell Bryant
........ r36725 | russell | 2006-07-03 00:19:09 -0400 (Mon, 03 Jul 2006) | 4 lines use ast_set_callerid to be more consistent and to make sure that the "callerid" option in the conf files is always handled the same way and sets ANI (issue #7285, gkloepfer) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36726 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-23revert my changes that converted the jb on the channel to be dynamicallyRussell Bryant
allocated. These changes caused crashes when using a channel type that did not support the jitterbuffer. Instead of fixing why it's crashing, I'm going to implement this in a better way next week. The way I did it caused a jitterbuffer to be allocated on every channel where the channel type supported jitterbuffers, even if they were disabled. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@35746 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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