[asterisk-bugs] [Asterisk 0013569]: Asterisk sending the wrong codec on re-invite.
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Dec 17 17:49:15 CST 2008
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: mnicholson
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Project: Asterisk
Issue ID: 13569
Category: Channels/chan_sip/CodecHandling
Reproducibility: always
Severity: minor
Priority: normal
Status: assigned
Asterisk Version: 1.6.0-rc6
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.1
SVN Revision (number only!):
Disclaimer on File?: N/A
Request Review:
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Date Submitted: 2008-09-26 17:48 CDT
Last Modified: 2008-12-17 17:49 CST
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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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(0096598) mnicholson (administrator) - 2008-12-17 17:49
http://bugs.digium.com/view.php?id=13569#c96598
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I reviewed the code that is supposed to handle this and I didn't see any
obvious problems. Would it be possible to get sip debug and asterisk debug
output of this issue so I can see some more of what asterisk is doing? I
will also try to reproduce this here using asterisk and openser.
Issue History
Date Modified Username Field Change
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2008-12-17 17:49 mnicholson Note Added: 0096598
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