[asterisk-bugs] [Asterisk 0013569]: Asterisk sending the wrong codec on re-invite.

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=13569 
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Reported By:                bkw918
Assigned To:                dvossel
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Project:                    Asterisk
Issue ID:                   13569
Category:                   Channels/chan_sip/CodecHandling
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     assigned
Asterisk Version:           1.6.2.0-beta1 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.1 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2008-09-26 17:48 CDT
Last Modified:              2009-04-14 15:34 CDT
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Summary:                    Asterisk sending the wrong codec on re-invite.
Description: 
FreeSWITCH sends invite out to tf.voipmich.com with PCMU, PCMA, GSM.  The
call is answered and setup using GSM since its listed first in the Answer
we receive from Asterisk.  A re-invite promptly follows offering
GSM,PCMU,PCMA to which we 200 OK with ONLY GSM in the SDP in our 200 OK.  

Promptly there after Asterisk starts sending PCMU packets.  The re-invite
is considered a new Session Offer Answer and Asterisk ignores the Answer
and sends a media format not in the new Answer.  

Asterisk PBX SVN-branch-1.6.0-r140976-/trunk is what I can see in the SDP
from John's equipment.  He has disabled re-invites pending a fix for this.

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 (0103244) dvossel (administrator) - 2009-04-14 15:34
 http://bugs.digium.com/view.php?id=13569#c103244 
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Hey bkw918, I'm trying to bring this issue to completion.  I've taken a
good look at it and agree that its a problem, but I'm really blind to
exactly what is causing it.  So far we are guessing about what happens
behind Asterisk before the native bridge takes place, so we really don't
know why that end point is sending ULAW rather than GSM.  At this point I
really need access to a packet capture of the Asterisk box as well as the
sip.conf information for the peers involved.  You mentioned that you would
be willing to to work with John Lodden to try and retrieve some of that
information.  Is this still possible? 

Issue History 
Date Modified    Username       Field                    Change               
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2009-04-14 15:34 dvossel        Note Added: 0103244                          
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