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

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=12322 
====================================================================== 
Reported By:                rolek
Assigned To:                
====================================================================== 
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:              
====================================================================== 
Date Submitted:             03-28-2008 06:10 CDT
Last Modified:              04-29-2008 10:19 CDT
====================================================================== 
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.
======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
related to          0006240 [branch] Errors in support for SIP stri...
====================================================================== 

---------------------------------------------------------------------- 
 rolek - 04-29-08 10:19  
---------------------------------------------------------------------- 
chan_sip.c_12322_reinvite_after_hangup.patch solved the problem, but broke
calls in some cases.

chan_sip.c_12322_reinvite_after_hangup_2.patch was one of my test patches
which can be removed. Please ignore it.

chan_sip.c_12322_reinvite_after_hangup_3.patch uses a dedicated variable
to indicate the call is a reinvite being hung up. This fixes the problem in
a much better way, without breaking any calls (AFAIK). 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
04-29-08 10:19  rolek          Note Added: 0086134                          
======================================================================