[asterisk-bugs] [Asterisk 0011901]: bridging chan_h323 and chan_sip
noreply at bugs.digium.com
noreply at bugs.digium.com
Wed Apr 9 13:00:59 CDT 2008
The following issue has been ASSIGNED.
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http://bugs.digium.com/view.php?id=11901
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Reported By: pj
Assigned To: file
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Project: Asterisk
Issue ID: 11901
Category: Core/Channels
Reproducibility: always
Severity: minor
Priority: normal
Status: assigned
Asterisk Version: SVN
SVN Branch (only for SVN checkouts, not tarball releases): trunk
SVN Revision (number only!): 101746
Disclaimer on File?: N/A
Request Review:
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Date Submitted: 02-01-2008 10:08 CST
Last Modified: 04-09-2008 13:00 CDT
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Summary: bridging chan_h323 and chan_sip
Description:
even if I using same codec (disallow=all, allow=g729) on both channels, I
never see 'Packet2Packet bridging...' message, like in sip-sip, sip-skinny,
skinny-skinny bridging.
I think, that h323 is using same RTP subsystem, so Packet2Packet bridging
should be possible to avoid RTP go through asterisk core, like in other
cases.
also some unnecessary warnings about codecs are displayed when calling
h323->sip, reverse direction sip->h323 is without warnings.
since both my ends supports g729, I have no g729 installed and expecting,
that asterisk will do simple passthrough. In general it works, but imho, it
should work better, with Packet2Packet bridging and avoid warnings below.
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svnbot - 04-09-08 13:00
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Repository: asterisk
Revision: 113840
U trunk/channels/chan_h323.c
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r113840 | file | 2008-04-09 13:00:56 -0500 (Wed, 09 Apr 2008) | 4 lines
Enable enough RTP bridging to allow P2P to work.
(closes issue http://bugs.digium.com/view.php?id=11901)
Reported by: pj
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http://svn.digium.com/view/asterisk?view=rev&revision=113840
Issue History
Date Modified Username Field Change
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04-09-08 13:00 svnbot Checkin
04-09-08 13:00 svnbot Note Added: 0085222
04-09-08 13:00 svnbot Status feedback => assigned
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