| Summary: | ASTERISK-16387: Direct RTP failures | ||
| Reporter: | Trevor Peirce #2 (digitalc) | Labels: | |
| Date Opened: | 2010-07-16 18:29:38 | Date Closed: | 2011-06-07 14:00:39 |
| Priority: | Major | Regression? | No |
| Status: | Closed/Complete | Components: | Channels/chan_sip/General |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ( 0) reinvite.txt | |
| Description: | Related to bug ASTERISK-13370 This bug is written from the perspective of Asterisk B, as that is there the problem seems to be. A call comes from OpenSER to Asterisk A. Asterisk A calls Asterisk B and reinvites audio, apparently successfully. The caller dials an extension. Asterisk B calls an IP phone with directmedia enabled. The caller can hear the dialed party, but the dialed party cannot hear the caller. It appears that Asterisk B neglects to tell Asterisk A where to send the audio to once the IP phone has been answered. ****** ADDITIONAL INFORMATION ****** The following is a summary of the attached debug log. Server B at 192.168.77.226 is running Asterisk 1.6.2.9 INVITE A 192.168.77.230 -> B 192.168.77.226 audio to A 192.168.77.230:10566 OK B -> A audio to B 192.168.77.226:25902 INVITE A -> B audio to A 192.168.88.104:32876 OK B -> A audio to B 192.168.77.226:25902 The caller dials an extension which creates a new channel INVITE B -> C audio to B 192.168.88.104:32876 OK C -> B audio to C 192.168.66.182:62410 The caller can hear the dialed party, but the dialed party hears silence because B never told A where to send audio! Tear down then appears to happen fine. BYE C -> B OK B -> C BYE B -> A OK A -> B | ||
| Comments: | By: Leif Madsen (lmadsen) 2010-07-20 10:09:22 I think it'd also be beneficial to have the configuration for Asterisk A and B so this can be reproduced in a lab. By: Leif Madsen (lmadsen) 2010-07-29 14:28:08 You're probably onto something here, but we really do need to see the configuration here or we're going to spend hours attempting to reproduce your particular issue. If you can provide the necessary information to reproduce this then we can move this forward. Also a SIP debug of the call setup is likely useful as well. By: Paul Belanger (pabelanger) 2010-08-18 08:00:30 Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested. Further information can be found at http://www.asterisk.org/developers/bug-guidelines | ||