[asterisk-bugs] [Asterisk 0013076]: Re-Invite occurs eventhough the codecs are incompatible.
noreply at bugs.digium.com
noreply at bugs.digium.com
Tue Jul 15 06:33:11 CDT 2008
A NOTE has been added to this issue.
======================================================================
http://bugs.digium.com/view.php?id=13076
======================================================================
Reported By: ramonpeek
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 13076
Category: Channels/chan_sip/CodecHandling
Reproducibility: always
Severity: minor
Priority: normal
Status: new
Asterisk Version: 1.4.21
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Disclaimer on File?: N/A
Request Review:
======================================================================
Date Submitted: 07-15-2008 06:27 CDT
Last Modified: 07-15-2008 06:33 CDT
======================================================================
Summary: Re-Invite occurs eventhough the codecs are
incompatible.
Description:
Re-invite occurs eventhough the codecs are incompatible.
See these steps to reproduce;
Device A accepts/offers codecs g711 & g729a (AudioCodes Mediant 1000)
Peer A in Asterisk only supports codec g711a
Peer B in Asterisk only supports codec g729a
Device B accepts/offers codec g729a (Snom Phone)
A call from device A is routed through Asterisk to device B.
Device B answers and then Asterisk sends a re-invite without a codec!!?
But why, The codecs don't even match!
Asterisk then prints-out the CLI-Error:
"ERROR[31381]: chan_sip.c:12326 handle_response_invite: Got error on T.38
re-invite. Bad configuration. Peer needs to have T.38 disabled."
Note:
If canreinvite is set to no the problem obviously does not occur.
And if peer a is set to allow G711a AND G729a the problem also does not
occur.
======================================================================
----------------------------------------------------------------------
ramonpeek - 07-15-08 06:33
----------------------------------------------------------------------
See attached trace with Debug 5, verbose 3 and SIP Debug on.
IP address: 192.168.161.4 is AudioCodes Mediant 1000 (originator)
IP address: 192.168.161.100 is Asterisk PBX
IP address: 192.168.161.216 is Snom 320 Phone running 7.1.34 firmwware.
Issue History
Date Modified Username Field Change
======================================================================
07-15-08 06:33 ramonpeek Note Added: 0090273
======================================================================
More information about the asterisk-bugs
mailing list