[asterisk-bugs] [Asterisk 0016036]: Asterisk unable to bridge RTP when a peer server performs a call transfer when canreinvite is enabled

[Asterisk Bug Tracker]

The following issue requires your FEEDBACK. 
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https://issues.asterisk.org/view.php?id=16036 
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Reported By:                jehanzeb
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16036
Category:                   Features/Parking
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           Older 1.4 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-10-07 10:50 CDT
Last Modified:              2009-10-21 09:37 CDT
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Summary:                    Asterisk unable to bridge RTP when a peer server
performs a call transfer when canreinvite is enabled
Description: 
Hi, I am having problems with the following scenario.
the call process is setup as follows.

handset A <---> server A <----> asterisk <---> server B <---> handset B
and C

when a call is placed by handset A to handset B, all works fine until
handset B tries to transfer the call to handset C. At this point when the
server B  tries to update asterisk with the ip address to which asterisk
should route the RTP to. Asterisk acknowledges the change but does not
update handset A of the change. therefore after the call has been
transfered even though Handset A is able to hear handset C, Handset C
cannot hear handset A. I will attach the sip configuration, dialplan and a
sip debug file to illustrate my point. Please let me know if there is any
other information i should provide.

I am currently running  Asterisk 1.4.21.2~dfsg-3

Regards,
Jehanzeb

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 (0112524) lmadsen (administrator) - 2009-10-21 09:37
 https://issues.asterisk.org/view.php?id=16036#c112524 
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1.4.21 is quite an old issue. Please verify the same situation presents
itself in the most recent version of Asterisk, ideally from subversion:

svn co http://svn.asterisk.org/svn/asterisk/branches/1.4
asterisk-1.4-vanilla

There is a good change this has been fixed quite a while ago. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-10-21 09:37 lmadsen        Note Added: 0112524                          
2009-10-21 09:37 lmadsen        Status                   new => feedback     
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