[asterisk-bugs] [Asterisk 0012566]: SIP falis to understan NAT clients
noreply at bugs.digium.com
noreply at bugs.digium.com
Thu May 1 13:26:26 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: new
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: 05-01-2008 13:26 CDT
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Summary: SIP falis to understan NAT clients
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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oej - 05-01-08 13:26
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If you had read the bug guidelines you would see that this is an incomplete
bug report. You also have several settings that is not valid. Please start
by contacting someone on IRC to help you or use the asterisk-users mailing
list to get help. If you still think we have a bug, please follow the
instructions and file a new bug report or re-open this one.
Thank you.
Issue History
Date Modified Username Field Change
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05-01-08 13:26 oej Note Added: 0086265
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