[asterisk-bugs] [Asterisk 0012566]: SIP falis to understan NAT clients

[noreply at bugs.digium.com]

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:                oej
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Project:                    Asterisk
Issue ID:                   12566
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     assigned
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:              05-05-2008 08:50 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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---------------------------------------------------------------------- 
 blitzrage - 05-05-08 08:50  
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FYI: We tried this both with and without directrtpsetup on. The same thing
occurred with it off. 

Issue History 
Date Modified   Username       Field                    Change               
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05-05-08 08:50  blitzrage      Note Added: 0086401                          
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