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193 Commits

Author SHA1 Message Date
oana 47908cf938 Add option to stop execution of an outgoing call. 
Propagate that parameter in dumb channel, tone generator and wave file.
In SIP, simulate going through the whole SIP stack before stopping execution of the call.



git-svn-id: http://voip.null.ro/svn/yate@6401 acf43c95-373e-0410-b603-e72c3f656dc1
 2020-04-22 10:59:48 +00:00
paulc 6b2daa2d48 Added G.722 codec from the WebRTC code. 
git-svn-id: http://voip.null.ro/svn/yate@6363 acf43c95-373e-0410-b603-e72c3f656dc1
 2019-11-06 18:31:54 +00:00
marian cc4f86d64b Added Channel parameters list to be set in all Channel messages. Update it from routing and chan.control. 
git-svn-id: http://voip.null.ro/svn/yate@6339 acf43c95-373e-0410-b603-e72c3f656dc1
 2019-01-14 09:54:04 +00:00
paulc 669bb07e1f Resolved or silenced a number of compile warnings. 
git-svn-id: http://voip.null.ro/svn/yate@5891 acf43c95-373e-0410-b603-e72c3f656dc1
 2014-07-31 16:30:15 +00:00
paulc d56cecb40e Added copyright notices to sources and scripts missing them. 
Updated copyright notices for 2014.
Fixed end of lines in many files.


git-svn-id: http://voip.null.ro/svn/yate@5755 acf43c95-373e-0410-b603-e72c3f656dc1
 2014-02-05 11:42:17 +00:00
marian 18ae05c63c Added post dial delay timeout in outgoing channel started from 'maxpdd' routing parameter. It stops when status is called with 'ringing', 'progressing' or 'answered'. 
git-svn-id: http://voip.null.ro/svn/yate@5666 acf43c95-373e-0410-b603-e72c3f656dc1
 2013-10-11 12:46:20 +00:00
marian 2ab24ed2e2 Changed im.execute relay id. Removed im.route message relay. Route chat using call.route message. 
git-svn-id: http://voip.null.ro/svn/yate@5618 acf43c95-373e-0410-b603-e72c3f656dc1
 2013-08-13 07:54:00 +00:00
paulc 18e10bfd69 Changed license terms in each source file to reference an external file. 
git-svn-id: http://voip.null.ro/svn/yate@5609 acf43c95-373e-0410-b603-e72c3f656dc1
 2013-08-06 13:38:10 +00:00
marian 6eb6b64cd7 Change host direction if possible and re-negotiate the transport when receiving stream hosts from remote. 
git-svn-id: http://voip.null.ro/svn/yate@5497 acf43c95-373e-0410-b603-e72c3f656dc1
 2013-04-26 12:31:28 +00:00
marian 8f3daacf72 Change file transfer host direction when failed to connect to remote provided host. Signal it to remote party. 
git-svn-id: http://voip.null.ro/svn/yate@5492 acf43c95-373e-0410-b603-e72c3f656dc1
 2013-04-25 14:55:29 +00:00
marian aac2758f81 Wait for session initiate result to send stream host for file transfer session. 
git-svn-id: http://voip.null.ro/svn/yate@5482 acf43c95-373e-0410-b603-e72c3f656dc1
 2013-04-23 11:13:10 +00:00
marian 089242bfc5 Fixed bug: use driver mutex to reset session user data pointer. 
git-svn-id: http://voip.null.ro/svn/yate@5463 acf43c95-373e-0410-b603-e72c3f656dc1
 2013-04-12 15:00:31 +00:00
marian b3ea9ff124 Don't remove file path when starting a file transfer path. 
git-svn-id: http://voip.null.ro/svn/yate@5432 acf43c95-373e-0410-b603-e72c3f656dc1
 2013-04-12 10:16:45 +00:00
marian 09796b168b Fixed bug: always look for transport candidates in session accept. 
git-svn-id: http://voip.null.ro/svn/yate@5359 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-12-14 08:52:22 +00:00
marian b8e9f19e66 Make sure the user data pointer carried by handled message is a CallEndpoint one. 
git-svn-id: http://voip.null.ro/svn/yate@5287 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-10-01 11:00:30 +00:00
paulc f5230b501c Changed the handling of the "timeout" parameter so it (re)starts when call is answered. 
Added "maxcall" and "timeout" setting and handling to channels that lacked it.


git-svn-id: http://voip.null.ro/svn/yate@5282 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-09-25 09:39:49 +00:00
paulc b6d72c4fc5 Always use the RTP started flag, no matter in which version. 
git-svn-id: http://voip.null.ro/svn/yate@5230 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-08-17 23:08:21 +00:00
paulc 2d3c720bba Fixed Jingle audio content creation in a scenario with immediate answer. 
git-svn-id: http://voip.null.ro/svn/yate@5229 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-08-17 21:09:08 +00:00
paulc 8a5cf1c398 Reset Jingle audio content on answer only if it was an early media one. 
git-svn-id: http://voip.null.ro/svn/yate@5228 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-08-17 18:02:21 +00:00
oana afc7680016 Set allowed number of simultaneous calls from configuration files. If not set, default to the maxchans setting in yate.conf. 
git-svn-id: http://voip.null.ro/svn/yate@5227 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-08-17 13:29:31 +00:00
paulc 7fc2e11287 Add the "bitrate" as a generic payload parameter in addition to the attribute. 
On inbound contents read the bitrate from generic parameter in preference to attribute.
Add the bitrate to the iLBC formats as some clients use it a a type hint instead of ptime.


git-svn-id: http://voip.null.ro/svn/yate@5204 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-07-17 14:25:38 +00:00
paulc 8df140fa4b List incoming Jingle codecs in caller's preferred order if available. 
git-svn-id: http://voip.null.ro/svn/yate@5182 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-07-05 17:02:10 +00:00
paulc e67b049d7d Lock the Jingle channel while changing its formats list. 
git-svn-id: http://voip.null.ro/svn/yate@5181 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-07-05 11:25:18 +00:00
paulc 8ca9cbe95d Support changing formats from routing for inbound Jingle calls. 
git-svn-id: http://voip.null.ro/svn/yate@5180 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-07-05 11:09:13 +00:00
paulc 5b0907dd91 Overwrite the "direction" parameter added in Rev. 5121 in "chan.rtp" messages. 
git-svn-id: http://voip.null.ro/svn/yate@5168 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-07-02 09:49:25 +00:00
paulc 7e77fd2cfb Added possibility to track message progress through handlers. 
git-svn-id: http://voip.null.ro/svn/yate@5107 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-06-12 23:47:01 +00:00
marian 465049f3e2 Enqueue call.ringing when initiate stanza is confirmed. 
git-svn-id: http://voip.null.ro/svn/yate@5040 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-05-04 09:13:49 +00:00
marian e63f9c68ae Send ringing if ring with content flag is set and call is progressing with ealy media 
git-svn-id: http://voip.null.ro/svn/yate@5038 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-05-04 07:51:04 +00:00
marian e96ad8b84e Added flags controlling ringing for incoming channels. Optionally add session content in jingle ringing stanza. 
git-svn-id: http://voip.null.ro/svn/yate@5033 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-05-01 14:30:26 +00:00
marian 26a0280e35 Added google raw udp transport support for jingle version 1. 
git-svn-id: http://voip.null.ro/svn/yate@5031 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-04-30 09:47:59 +00:00
marian 778be7bbc3 Added support to offer p2p transport for outgoing jingle sessions. 
git-svn-id: http://voip.null.ro/svn/yate@4982 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-03-27 07:37:56 +00:00
marian 176817e50e Added isac to jingle known codecs. 
git-svn-id: http://voip.null.ro/svn/yate@4957 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-03-06 16:01:37 +00:00
marian 72458b0ddb Reset received rtp ssrc media attribute to avoid sending it back. 
git-svn-id: http://voip.null.ro/svn/yate@4901 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-02-13 10:04:07 +00:00
marian 87ca662604 Refuse new incoming calls if don't accepted by the driver (engine exiting or full) in sig, iax and jingle modules. 
git-svn-id: http://voip.null.ro/svn/yate@4794 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-01-04 15:58:31 +00:00
marian 60ea5b41c4 Fixed jingle SRTP negotiation and setup. 
git-svn-id: http://voip.null.ro/svn/yate@4727 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-11-29 15:51:14 +00:00
paulc 4d74b5138f Added new static mutex to protect channel disconnect parameters during access. 
git-svn-id: http://voip.null.ro/svn/yate@4662 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-10-28 18:23:26 +00:00
paulc 005c873bce Allow global calls limits to be reloaded. 
git-svn-id: http://voip.null.ro/svn/yate@4636 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-09-26 14:55:42 +00:00
marian 253c8006af Added google p2p transport support for jingle version 1. 
git-svn-id: http://voip.null.ro/svn/yate@4479 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-07-11 09:32:16 +00:00
paulc 07132042d0 Added class for pseudo random number generation. 
Replace library ::random() and ::rand() calls with shared Random method.


git-svn-id: http://voip.null.ro/svn/yate@4470 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-06-29 11:19:02 +00:00
marian 8f97fb50a6 Properly match ilbc when handling formats set from routing. 
git-svn-id: http://voip.null.ro/svn/yate@4366 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-05-11 17:11:25 +00:00
marian a928417682 Removed unused function. 
git-svn-id: http://voip.null.ro/svn/yate@4308 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-04-18 08:35:31 +00:00
marian 5dec2bf000 Conditionally compile function used only for debug purposes. 
git-svn-id: http://voip.null.ro/svn/yate@4307 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-04-18 08:32:39 +00:00
marian d99f89a63e Fixed bug: restart rtp and transport when used media format or payload id changes in received session accept. Added debug. 
git-svn-id: http://voip.null.ro/svn/yate@4297 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-04-14 14:41:33 +00:00
marian 24d0606696 Handle (o)jingle_flags parameter from routing and replace session flags. Set the parameter when redirecting. 
git-svn-id: http://voip.null.ro/svn/yate@4281 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-04-11 14:50:17 +00:00
marian 2375867d9b Initialize jingle engine when module initializes. 
git-svn-id: http://voip.null.ro/svn/yate@4279 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-04-11 14:17:50 +00:00
marian 32a7a8d36c Remember telephone event payload id and name for received contents and reflect them when responding. Update telephone event for sent contents. Set event payload id when starting rtp. 
git-svn-id: http://voip.null.ro/svn/yate@4246 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-03-31 11:16:36 +00:00
marian 677666e486 Fixed media checking for incoming contents. Fixed ilbc codec negotiation. Added speex to codecs list. 
git-svn-id: http://voip.null.ro/svn/yate@4240 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-03-30 15:14:40 +00:00
marian 5ca00bf96e Set yate node in presence entity capability child. 
git-svn-id: http://voip.null.ro/svn/yate@4217 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-03-25 10:31:09 +00:00
marian 91539fd0d4 Allow overriding dtmf method on incoming jingle channels. 
git-svn-id: http://voip.null.ro/svn/yate@4205 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-03-24 09:59:14 +00:00
marian 9b3844a03e Avoid adding twice the module parameter when dispatching jabber.account messages. 
git-svn-id: http://voip.null.ro/svn/yate@4201 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-03-23 14:03:48 +00:00