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