[Asterisk-Users] Confused about NAT and Authentication with FWD

[Kristian Kielhofner]

ian at drumoak.demon.co.uk wrote:
> I have recently started experimenting with Asterisk. I am running the
> system the other side of the a NAT router and trying to connect to
> FWD. I have opened UDP ports and have configured sip.conf to handle
> NAT.
> 
> The problem:
> 
> I can call from the FWD phone and the extension on Asterisk rings and
> there is two way sound so no problem.
> 
> Now if in the extension.conf file I have, exten =>
> _.,3,Dial(SIP/${EXTEN}@fwd,20)
> 
> where fwd is the same name as the definition in sip.conf then I can
> see that Asterisk is handling the call as if it is going across the
> NAT ( see attached sip debug output). However (there is always a
> however) I get  and the call fails.
> 
> Oct  6 20:01:20 NOTICE[1087269568]: chan_sip.c:6766 handle_response:
> Failed to authenticate on INVITE to '"465605"
> <sip:465605 at fwd.pulver.com>;tag=as1ba1c908'
> 
> If I change the Dial entry in extension.conf to
> 
> exten => _.,3,Dial(SIP/${EXTEN}@fwd.pulver.com,20)
> 
> Then I no longer see the outgoing connection being handled as if it
> is going across a NAT. The call does connect but there is only sound
> from the asterisk originating end I cannot get sound from the FWD
> end.
> 
> What Am I doing wrong.
> 

Two things to do that I can think of:

1) In sip.conf, add externip="your ip here" under [general].  "Your ip 
here" should be the public address that you connect to the net with.  If 
you are on the same network as the * box, go to getip.dyndns.org - and 
it will tell you what to put here.

2) Modify rtp.conf to change the port range, and then tell your NAT 
firewall to forward that range (as well as SIP) to your * box.

	Other than that, it is FWD specific...  You may want to look at 
chan_sip2 (which has outgoing proxy support, something that FWD uses to 
help with the SIP NAT problem).

--
Kristian Kielhofner