[asterisk-bugs] [Asterisk 0013076]: Re-Invite occurs eventhough the codecs are incompatible.

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=13076 
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Reported By:                ramonpeek
Assigned To:                
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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:              
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Date Submitted:             2008-07-15 06:27 CDT
Last Modified:              2008-09-07 23:11 CDT
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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.

 
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 (0092164) samdell3 (reporter) - 2008-09-07 23:11
 http://bugs.digium.com/view.php?id=13076#c92164 
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Ive seen this error before. It usually happens when you try setting
SIP_CODEC before Dial(). The error message is generated becasue the CPE
sends back a 488 to asterisk after the re-invite (that should never have
happened anyway).

Are you using SIP_CODEC?  If so, see bug 13243 to fix it.

If you are not using SIP_CODEC then there must be something else really
broken with v1.4.21 

Issue History 
Date Modified    Username       Field                    Change               
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2008-09-07 23:11 samdell3       Note Added: 0092164                          
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