[asterisk-bugs] [Asterisk 0012437]: Asterisk negotiates only T.38 when answering even if the other end offers audio

Asterisk Bug Tracker noreply at bugs.digium.com
Wed Mar 4 17:21:50 CST 2009


A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=12437 
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Reported By:                marsosa
Assigned To:                file
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Project:                    Asterisk
Issue ID:                   12437
Category:                   Channels/chan_sip/T.38
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     ready for testing
Asterisk Version:           1.4.18 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2008-04-14 08:31 CDT
Last Modified:              2009-03-04 17:21 CST
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Summary:                    Asterisk negotiates only T.38 when answering even if
the other end offers audio
Description: 
One of our gateways (audiocodes mp-118) offers ulaw,g729 and t.38 when an
incoming call is sent to asterisk, and asterisk answer() with t.38 only,
instead of using ulaw. T.38 is enabled on the gateway because this is
needed for reinvites, if i disable it, the call works ok but fails later
when the ata wants to do reinvite for receiving faxes with t.38 '488 not
acceptable'.
The main problem here is that, after answering with t.38, asterisk sends
invites with t.38 only to the ip phones, and they rejected with not
acceptable.
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---------------------------------------------------------------------- 
 (0101236) pinga-fogo (reporter) - 2009-03-04 17:21
 http://bugs.digium.com/view.php?id=12437#c101236 
---------------------------------------------------------------------- 
this sip trace is ok... the problem is i receive a call from gvt but
asterisk does not invite the phones 
[Mar 4 11:07:07] WARNING[27302]: chan_sip.c:3081 sip_call: No audio format
found to offer. Cancelling call to 5002

i can see ane invite from asterisk to the phones... try this to reproduce
the issue...

create a peer with t38(with the initial invite) and g711 support...
and create a peer with only g711 suppor and try to do a voice call... 
i post my configs latter ok.. 
thanks 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-03-04 17:21 pinga-fogo     Note Added: 0101236                          
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