|Summary:||ASTERISK-21045: Session refresh reinvites an in progress T.38 dialog back to G.711|
|Reporter:||Trevor Peirce (trev)||Labels:|
|Date Opened:||2013-02-06 22:18:09.000-0600||Date Closed:||2013-04-10 09:15:13|
|Environment:||Attachments:||( 0) t38_session_timer.pcap|
|Description:||A call is initiated and successfully reinvited to T.38. Both ends are happily communicating. Once the session timer runs out, Asterisk sends an invite but this invite changes the audio stream back to G.711. It only does this through one channel, so we end up with the caller sending T.38 and the callee sending G.711. Things spiral downhill from there.
End result is faxes that take longer than the session timer are not able to complete.
|Comments:||By: Matt Jordan (mjordan) 2013-02-13 17:20:26.618-0600|
Just so we have a clear picture of the call flow, can you attach a pcap demonstrating the re-INVITE back to audio?
By: Trevor Peirce (trev) 2013-02-13 17:47:44.468-0600
Here's a capture of the SIP signalling for both legs of the call.
By: Rusty Newton (rnewton) 2013-02-15 15:31:40.533-0600
Thanks Trevor. Yup, looks buggy. Acknowledging.
By: Trevor Peirce (trev) 2013-04-10 02:05:51.958-0500
21232 is a duplicate of this.
By: David Brillert (aragon) 2013-04-10 08:38:43.583-0500
Trevor, if this is an exact duplicate of 21232 that issue was fixed and the changes committed in 11.4.0-rc1
2013-03-15 01:34 +0000 [r383121-383125] Matthew Jordan <email@example.com>
* /, channels/chan_sip.c: When a session timer expires during a
T.38 call, re-invite with correct SDP When a session timer
expires during a dialog that has re-negotiated to T.38 and
Asterisk is the refresher, Asterisk will send a re-INVITE with an
SDP containing audio media only. This causes some hilarity with
the poor fax session under weigh. This patch corrects that by
sending T.38 parameters if we are in the middle of a T.38
session. (closes issue ASTERISK-21232) Reported by: Nitesh Bansal
uploaded by nbansal (License 6418) ........ Merged revisions
383124 from http://svn.asterisk.org/svn/asterisk/branches/1.8
By: Matt Jordan (mjordan) 2013-04-10 09:15:13.437-0500
Thanks David for pointing that out. I'm going to go ahead and close this out as a duplicate of that issue - Trevor, if you find that the issue isn't resolved in Asterisk 11.4.0-rc1, please let a bug marshal know in #asterisk-bugs and we'll be happy to reopen the issue. Thanks!