[asterisk-bugs] [Asterisk 0015545]: Not passing audio on a sip call in and out on the same peer
Asterisk Bug Tracker
noreply at bugs.digium.com
Fri Jun 18 10:01:56 CDT 2010
The following issue has been CLOSED
======================================================================
https://issues.asterisk.org/view.php?id=15545
======================================================================
Reported By: kobaz
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 15545
Category: Channels/chan_sip/General
Reproducibility: always
Severity: major
Priority: normal
Status: closed
Asterisk Version: 1.6.0.10
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: 2009-07-21 10:07 CDT
Last Modified: 2010-06-18 10:01 CDT
======================================================================
Summary: Not passing audio on a sip call in and out on the
same peer
Description:
This worked in 1.4.x, so I'm assuming this is a bug.
Call comes in from an itsp via sip. We then proceed to dial out that same
itsp (ie: call forwarding). The remote side answers the call, but no audio
is passed.
This happens on 1.6.0.10, but it's not available as a product version.
rtp packets are zero during the call. The asterisk box is also not behind
nat.
======================================================================
----------------------------------------------------------------------
(0123580) pabelanger (manager) - 2010-06-18 10:01
https://issues.asterisk.org/view.php?id=15545#c123580
----------------------------------------------------------------------
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
Issue History
Date Modified Username Field Change
======================================================================
2010-06-18 10:01 pabelanger Note Added: 0123580
2010-06-18 10:01 pabelanger Status feedback => closed
======================================================================
More information about the asterisk-bugs
mailing list