[asterisk-bugs] [Asterisk 0017666]: Direct RTP failures
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Aug 18 08:00:31 CDT 2010
The following issue has been UPDATED.
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https://issues.asterisk.org/view.php?id=17666
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Reported By: digitalc
Assigned To:
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Project: Asterisk
Issue ID: 17666
Category: Channels/chan_sip/General
Reproducibility: always
Severity: major
Priority: normal
Status: closed
Asterisk Version: 1.6.2.9
JIRA: SWP-1869
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
Resolution: suspended
Fixed in Version:
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Date Submitted: 2010-07-16 18:29 CDT
Last Modified: 2010-08-18 08:00 CDT
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Summary: Direct RTP failures
Description:
Related to bug https://issues.asterisk.org/view.php?id=14244
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.
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Relationships ID Summary
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related to 0014244 No Audio on Call Transfer (Invite not b...
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(0126085) pabelanger (manager) - 2010-08-18 08:00
https://issues.asterisk.org/view.php?id=17666#c126085
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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
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Issue History
Date Modified Username Field Change
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2010-08-18 08:00 pabelanger Note Added: 0126085
2010-08-18 08:00 pabelanger Status feedback => closed
2010-08-18 08:00 pabelanger Resolution open => suspended
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