[asterisk-bugs] [Asterisk 0019010]: Call stuck with "music on hold"

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=19010 
====================================================================== 
Reported By:                ovi
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   19010
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.8.3.2 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2011-03-22 15:03 CDT
Last Modified:              2011-03-23 14:18 CDT
====================================================================== 
Summary:                    Call stuck with "music on hold"
Description: 
We have a simple call scenario: caller calls callee and callee puts caller
on hold.  The caller receives music on hold.
When the callee tries to resume the call, music on hold is still being
played to the caller.  What is different from a regular scenario here is
the fact that the port and IP in SDP in the resume reINVITE is not the same
as it was before (when the media was active).
====================================================================== 

---------------------------------------------------------------------- 
 (0133096) ovi (reporter) - 2011-03-23 14:18
 https://issues.asterisk.org/view.php?id=19010#c133096 
---------------------------------------------------------------------- 
The "ignoresdpversion=yes" did the trick (thanks davidw for pointing that
out).

Just for the record, here's the scenario that I was testing:
The callee is an opensips server which provides SCA (shared call
appearance) to two SIP phones (the .92 and the .93).  Opensips was acting
as a signalling B2BUA and passed SDP unchanged from the two phones to
asterisk (SDP1 and SDP2 were belonging to phoneA, and when the call was
retrieved by phoneB, the SDP3 from phoneB was sent to asterisk, hence the
change in session version).

This bug can be closed now as a workaround has been identified. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2011-03-23 14:18 ovi            Note Added: 0133096                          
======================================================================