| 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 |
| Priority: | Minor | Regression? | No |
| Status: | Closed/Complete | Components: | Features/Parking |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| 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 1.4.21.2~dfsg-3 Regards, Jehanzeb | ||
| Comments: | By: Jehanzeb Mansoor (jehanzeb) 2009-10-20 10:53:53 Hi Guys, Any ideas for this one yet Regards, Jehanzeb 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. | ||