[asterisk-bugs] [Asterisk 0012322]: SIP reinvite record-route problem after hangup

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=12322 
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Reported By:                rolek
Assigned To:                
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Project:                    Asterisk
Issue ID:                   12322
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.18 
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:             03-28-2008 06:10 CDT
Last Modified:              07-01-2008 09:06 CDT
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Summary:                    SIP reinvite record-route problem after hangup
Description: 
Situation: phone1 - *a - *b - provider - phone2

When making a call from phone2 to phone1, both *b and provider use
re-invites to get out of the RTP stream. After phone1 hangs up, *b tries to
send BYE directly to the RTP server of the provider instead of its SIP
peer. The result is that phone2 does not see that the call has ended.
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Relationships       ID      Summary
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related to          0006240 [branch] Errors in support for SIP stri...
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 oej - 07-01-08 09:06  
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I need a full debug with verbose and everything you get at the console. I
don't see the issue here as I don't get all information. Please upload the
output of all logging channels, not only the debug channel, and from the
start of the call to the failed bye. Thank you. 

Issue History 
Date Modified   Username       Field                    Change               
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07-01-08 09:06  oej            Note Added: 0089497                          
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