Summary:ASTERISK-14955: Asterisk unable to bridge RTP when a peer server performs a call transfer when canreinvite is enabled
Reporter:Jehanzeb Mansoor (jehanzeb)Labels:
Date Opened:2009-10-07 10:50:34Date Closed:2011-06-07 14:08:22
Versions:Frequency of
Environment:Attachments:( 0) call_transer.txt
( 1) dialplan.txt
( 2) peers.txt
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

Comments:By: Jehanzeb Mansoor (jehanzeb) 2009-10-20 10:53:53

Hi Guys,

Any ideas for this one yet


By: Leif Madsen (lmadsen) 2009-10-21 09:37:23

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.

By: Leif Madsen (lmadsen) 2009-12-21 09:34:17.000-0600

Closed due to lack of feedback from reporter.