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 22:18:09.000-0600Date Closed:2013-04-10 09:15:13
Versions:11.2.0 Frequency of
duplicatesASTERISK-21232 Asterisk sends AUDIO REINVITE when session timer expires in T38 call
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 <mjordan@digium.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!