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

Asterisk Bug Tracker noreply at bugs.digium.com
Mon Dec 21 09:34:17 CST 2009


The following issue has been CLOSED 
====================================================================== 
https://issues.asterisk.org/view.php?id=16036 
====================================================================== 
Reported By:                jehanzeb
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   16036
Category:                   Features/Parking
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     closed
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:              
Resolution:                 open
Fixed in Version:           
====================================================================== 
Date Submitted:             2009-10-07 10:50 CDT
Last Modified:              2009-12-21 09:34 CST
====================================================================== 
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

====================================================================== 

---------------------------------------------------------------------- 
 (0115527) lmadsen (administrator) - 2009-12-21 09:34
 https://issues.asterisk.org/view.php?id=16036#c115527 
---------------------------------------------------------------------- 
Closed due to lack of feedback from reporter. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-12-21 09:34 lmadsen        Note Added: 0115527                          
2009-12-21 09:34 lmadsen        Status                   feedback => closed  
======================================================================




More information about the asterisk-bugs mailing list