|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 10:50:34||Date Closed:||2011-06-07 14:08:22|
|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 126.96.36.199~dfsg-3
|Comments:||By: Jehanzeb Mansoor (jehanzeb) 2009-10-20 10:53:53|
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.