[asterisk-bugs] [Asterisk 0012566]: Loss of audio after 2 seconds of P2P RTP bridging
noreply at bugs.digium.com
noreply at bugs.digium.com
Wed May 7 10:44:58 CDT 2008
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=12566
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Reported By: falves11
Assigned To:
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Project: Asterisk
Issue ID: 12566
Category: Channels/chan_sip/General
Reproducibility: always
Severity: major
Priority: normal
Status: feedback
Asterisk Version: SVN
SVN Branch (only for SVN checkouts, not tarball releases): trunk
SVN Revision (number only!): 114912
Disclaimer on File?: N/A
Request Review:
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Date Submitted: 05-01-2008 13:23 CDT
Last Modified: 05-07-2008 10:44 CDT
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Summary: Loss of audio after 2 seconds of P2P RTP bridging
Description:
I have a peer like this
[federico]
type=peer
host=72.187.243.97
context=default
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=g729
allow=ulaw
nat=yes
insecure=port,invite
but when I type "sip show peers"
ame/username Host Dyn Nat ACL Port Status
federico 72.187.243.97 N 5060
Unmonitored
and in fact the calls connect and there is second of audio and then the
audio stops. I am behind a nat.
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falves11 - 05-07-08 10:44
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Is there a volunteer that wishes to help capture the debugging information,
since Digium's Leif has no time and I don't feel like paying support for
bugs? I can dedicate a machine only for analizing this bug.
Issue History
Date Modified Username Field Change
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05-07-08 10:44 falves11 Note Added: 0086539
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