[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
Thu Mar 26 10:57:17 CDT 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:              
====================================================================== 
Date Submitted:             2008-04-14 08:31 CDT
Last Modified:              2009-03-26 10:57 CDT
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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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---------------------------------------------------------------------- 
 (0102238) afu (reporter) - 2009-03-26 10:57
 http://bugs.digium.com/view.php?id=12437#c102238 
---------------------------------------------------------------------- 
hello again!

reinvite to no on all peers doen't change anything (and i think, t38
wouldn't go any longer, because it would need reinvite, if i'm right...)

did more tests:
2 Sides Teles Gateways direct connect works fine.
proxy through asterisk has funny options.

phone rings, b takes the call. a hears b because of inband info, gets no
connect! of cause its an one way audio.
after about a minute a is released by asterisk.
if b drops the line within about 30 seconds the call leg to a is setup
again!
(this part is not in the trace file)

see attached files 
teles ast teles calling  (asttrace).txt 
and 
teles ast teles called  (asttrace).txt

greetings andy 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-03-26 10:57 afu            Note Added: 0102238                          
======================================================================




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