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

Asterisk Bug Tracker noreply at bugs.digium.com
Mon May 18 08:57:33 CDT 2009


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=13569 
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Reported By:                bkw918
Assigned To:                file
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Project:                    Asterisk
Issue ID:                   13569
Category:                   Channels/chan_sip/CodecHandling
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     closed
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:              
Resolution:                 fixed
Fixed in Version:           
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Date Submitted:             2008-09-26 17:48 CDT
Last Modified:              2009-05-18 08:57 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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---------------------------------------------------------------------- 
 (0104926) svnbot (reporter) - 2009-05-18 08:57
 https://issues.asterisk.org/view.php?id=13569#c104926 
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Repository: asterisk
Revision: 195097

_U  branches/1.6.0/
U   branches/1.6.0/main/rtp.c

------------------------------------------------------------------------
r195097 | file | 2009-05-18 08:57:33 -0500 (Mon, 18 May 2009) | 19 lines

Merged revisions 195096 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

................
  r195096 | file | 2009-05-18 10:56:16 -0300 (Mon, 18 May 2009) | 12 lines
  
  Merged revisions 195095 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r195095 | file | 2009-05-18 10:53:39 -0300 (Mon, 18 May 2009) | 5
lines
    
    Fix a bug where the codecs of the called party leg were not properly
sent back to the caller call leg when reinvited.
    
    (closes issue https://issues.asterisk.org/view.php?id=13569)
    Reported by: bkw918
  ........
................

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http://svn.digium.com/view/asterisk?view=rev&revision=195097 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-05-18 08:57 svnbot         Checkin                                      
2009-05-18 08:57 svnbot         Note Added: 0104926                          
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