[asterisk-users] SIP & NAT

[Mike Hammett]

If I have several local networks, can I specify that?

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Eric
"ManxPower" Wieling
Sent: Thursday, March 29, 2007 1:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP & NAT

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.
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