[asterisk-users] SIP & NAT

[Eric "ManxPower" Wieling]

Mike Hammett wrote:
> I hate SIP.  The only reason I'm doing this is that its cheaper than
> deploying the server to a colo facility.  My provider has given me a
> non-standard IP block, so I can't do typical routing.
> 
>  
> 
> I have Asterisk server <-> MT w\NAT <-> PPPoE <-> MT <-> Provider.
> 
>  
> 
> I setup a dst-nat on 5060 to the Asterisk box.
> 
>  
> 
> Audio from Asterisk -->  PSTN works great.  Audio Asterisk <-- PSTN does
> not.

That would be expected since you did not forward the ports used for RTP. 
  See /etc/asterisk/rtp.conf  A sample is in the Asterisk source.

Did you also set localnet= and externip= options in sip.conf [general].

SIP works just fine with NAT if you have it correctly configured and 
your server is on a static IP address.