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

noreply at bugs.digium.com noreply at bugs.digium.com
Sun May 4 22:56:43 CDT 2008


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=12566 
====================================================================== 
Reported By:                falves11
Assigned To:                oej
====================================================================== 
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:              
====================================================================== 
Date Submitted:             05-01-2008 13:23 CDT
Last Modified:              05-04-2008 22:56 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.
====================================================================== 

---------------------------------------------------------------------- 
 falves11 - 05-04-08 22:56  
---------------------------------------------------------------------- 
After the call connects to the destination, and I hangup in the
destination, I see these errors on the screen. All this happens if the
originating SIP endpoint is behind a NAT and there is no DMZ active.
Somehow the way we handle NAT got lost from 1.4 to Trunk.

-- Packet2Packet bridging SIP/45990-b805d4f8 and
SIP/4.71.122.130-b80b7568
Sipserver208*CLI> [May  4 23:57:54] ERROR[3500]: chan_sip.c:19213
handle_request_do: We could NOT get the channel lock for
SIP/45990-b805d4f8!
[May  4 23:57:54] ERROR[3500]: chan_sip.c:19214 handle_request_do: SIP
transaction failed: 0B87AA19-3AF6-4238-A858-70A8F5204419 at 192.168.1.111
[May  4 23:57:54] ERROR[3500]: chan_sip.c:19213 handle_request_do: We
could NOT get the channel lock for SIP/45990-b805d4f8!
[May  4 23:57:54] ERROR[3500]: chan_sip.c:19214 handle_request_do: SIP
transaction failed: 0B87AA19-3AF6-4238-A858-70A8F5204419 at 192.168.1.111 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
05-04-08 22:56  falves11       Note Added: 0086394                          
======================================================================




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