[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
Wed Jun 2 13:40:28 CDT 2010
The following issue requires your FEEDBACK.
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https://issues.asterisk.org/view.php?id=15545
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Reported By: kobaz
Assigned To:
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Project: Asterisk
Issue ID: 15545
Category: Channels/chan_sip/General
Reproducibility: always
Severity: major
Priority: normal
Status: feedback
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:
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Date Submitted: 2009-07-21 10:07 CDT
Last Modified: 2010-06-02 13:40 CDT
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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.
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(0122826) pabelanger (manager) - 2010-06-02 13:40
https://issues.asterisk.org/view.php?id=15545#c122826
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Is this still an issue using 1.6.2? Also, try using Progress() in you
dialplan.
---
Per the Asterisk maintenance timeline page at
http://www.asterisk.org/asterisk-versions maintenance (bug) support for the
1.6.0 and 1.6.1 branches has ended. For continued maintenance support
please move to the 1.6.2 branch.
More information on this change can be found in the release announcement:
http://www.asterisk.org/node/49924
Issue History
Date Modified Username Field Change
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2010-06-02 13:40 pabelanger Note Added: 0122826
2010-06-02 13:40 pabelanger Status acknowledged =>
feedback
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