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

noreply at bugs.digium.com noreply at bugs.digium.com
Thu May 1 17:35:04 CDT 2008


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=12566 
====================================================================== 
Reported By:                falves11
Assigned To:                
====================================================================== 
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:              
====================================================================== 
Date Submitted:             05-01-2008 13:23 CDT
Last Modified:              05-01-2008 17:35 CDT
====================================================================== 
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.
====================================================================== 

---------------------------------------------------------------------- 
 blitzrage - 05-01-08 17:35  
---------------------------------------------------------------------- 
On this system with no changes to the configuration, I was able to see this
issue reproduced.

What happens on 1.4 is that the call is setup to my IVR, and audio is
passed in both directions (confirmed with 'rtp set debug on' and that fact
we could hear each other).

When running trunk, the system seems to pass RTP for a few seconds, but
then stops. No audio is heard.

I have provided the console output. My apologies for the useless
information at the top of the file -- I'm running a bit late and didn't
have time to format it nicely for you.

Please let me know if you need any additional information to move this
forward.

Thanks!
L. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
05-01-08 17:35  blitzrage      Note Added: 0086293                          
======================================================================




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