asterisk

Commit Graph

Author	SHA1	Message	Date
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
Joshua Colp	354c03c4e6	Update chan_jingle to new definition of set_rtp_peer. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@55088 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	ac4090fce0	add another dependency git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53785 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
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	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	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
Matt O'Gorman	05a695af72	everything that loads a config that needs a config file to run 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	19 years ago
Joshua Colp	c6977b9983	Merge in VLDTMF support with Zaptel/Core done by the ever great Darumkilla 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	19 years ago
Russell Bryant	6aae631cc9	update to reflect recent rtp changes git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41272 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Kevin P. Fleming	0a27d8bfe5	merge new_loader_completion branch, including (at least): - 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	19 years ago
Russell Bryant	9f9a5f1984	move the calls to ast_jb_configure() to before the PBX thread is started on the 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	19 years ago
Matt O'Gorman	1ef09ebfed	some code clean up and catch for a act_hook being called without a packet. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39351 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Matt O'Gorman	3f115f8c31	Many many code cleanup changes given to me by Oej 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	19 years ago
Matt O'Gorman	a8d7d9123a	dtmf support. not everything else, trying to clear out those other bugs but more to come i guess. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38714 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	ca9ba719b6	Merge a new implementation of ast_inet_ntoa, our thread safe replacement for 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	19 years ago
Kevin P. Fleming	6d0742fc16	merge Russell's 'hold_handling' branch, finally implementing music-on-hold 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	19 years ago
Kevin P. Fleming	fd9c9ec28f	allow users of RTP to use G726-32 AAL2 packing even when RFC3551 packing has 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	19 years ago
Russell Bryant	73e8e2ab1f	Blocked revisions 36725 via svnmerge ........ 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	19 years ago
Russell Bryant	c8ceb92a4f	revert my changes that converted the jb on the channel to be dynamically 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	19 years ago
Russell Bryant	46018d5032	- dynamically allocate the ast_jb structure that is on the channel structure so that channels not using a jitterbuffer don't waste as much memory - ensure that the channel drivers that use jitterbuffers can handle a failure from configuring a jitterbuffer on a new channel because of a memory allocation error - On passing through these channel drivers, configure the jitterbuffer before starting the PBX thread instead of afterwards. If the pbx fails to start for whatever reason, this would have caused a crash. - Also on passing, move the increase of the usecount to after all of the possible failure conditions in the function - fix a place where ast_update_use_count() was not called - ensure that the owner channel pointer of the channel pvt strcutures is set to NULL in failure conditions git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@35553 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	f75ad9736a	bail if ast_calloc fails, this was done before but i accidently removed it when moving these allocations so duplicate error messages were not produced (issue #7345) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@34663 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	7ec13047bc	fix various coding guidelines issues (issue #7345 , with additional changes) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@34631 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Kevin P. Fleming	472c1ca282	simplify autoconfig include mechanism (make tholo happy he can use lint again :-) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32846 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	4c76028de9	- add the ability to configure forced jitterbuffers on h323, jingle, and mgcp channels - remove the jitterbuffer configuration from the pvt structures in the sip, zap, and skinny channel drivers, as copying the same global configuration into each pvt structure has no benefit. - update and fix some typos in jitterbuffer related documentation (issue #7257, north, with additional updates and modifications) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31413 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	0384330d64	update the rest of the channel drivers that use RTP so that their channel tech structures indicate that they create jitter git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31077 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Tilghman Lesher	705b5459c1	Bug 7237 - Replace recoded thread_safe_rand with the existing ast_random API git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@30906 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	1203424ce2	remove duplicate static keywords, oops git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@30723 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	1617adb055	make some variables static ... committed from xcode :) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@30655 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Mark Spencer	ec42807ebd	That goes for jingle too :) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@30521 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Matt O'Gorman	5d51260c36	finish cleaning up some more stuff before russell gets a chance to. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29708 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	32a28ed9cf	update chan_jingle to reflect the recent change to the indicate prototype git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29703 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Matt O'Gorman	7aa1a77e75	asterisk-xmpp merge in git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29553 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago

37 Commits (0fb8d82f18ee154fc85228fda85a51365e6eed47)