[Asterisk-Users] Creating a SIP gateway for use behind NAT

[Leif Madsen]

Leif Madsen wrote:

> OK.. lets just simplify this a bit.
> 
> <remote> <--TDM400P--> <*> <--IAX--> <*> <--SIP--> <remote>
> 
> I want to make a call from the very left asterisk box, through the right 
> asterisk box to the remote end.  So I have a phone plugged into the left 
> * box, which is connected via an IAX connection to the right * box, 
> which then places the call for me over SIP.  I figure this should allow 
> me to traverse NAT if the right * box is also on the GW machine (which I 
> know works with SIP as it's just a firewall issue, and not NAT, which is 
> simple).
> 
> So, if someone can help explain how I can make a call from the left * 
> box, through the right * box to the remote SIP connection, I'd be very 
> grateful!  I also promise to document anything and post it to the list 
> for archival purposes, and on a website for an online reference.

Simply for archival purposes, this is now working.  You can find the 
.conf's at http://www.hacklocalhost.com/asterisk.php

Thanks,
Leif Madsen.