Actually

Mark Phillips wrote:
> This advice is WRONG.
> 
> You not only need to forward port 5060 but also your RTP ports as well
> (10K to 20K by AAH default). Without any RTP you'll get a call setup
> correctly but no audio will flow.
> 
> When using IAX2 you only need forward port 4569 to your Asterisk server.
> You should check with your ITSP which version of IAX he supports. It's
> probably IAX2 (been around since Asterisk V1) but it could be IAX if
> he's on older software (pre V1).
> 

I use Asterisk fine without forwarding the RTP ports. But I also have 
mine setup for use with the externIP option. You also have to turn off 
reinvite on your SIP extensions. If you don't turn off reinvite on your 
sip extension they will attempt to bypass Asterisk with the media 
stream. I guess you could leave it on if you setup a different port 
range to forward for each device and set the device to only use that 
range for RTP.

Asterisk really only works in this setup if both the clients(extensions) 
and the server are behind the same NAT. Which is my case. If the 
extensions are outside the NAT they must be public IP's. My setup I have 
  asterisk and two SIP phones behind M0n0wall. Then I connect to 2 SIP 
providers. I also connect to two IAX2 providers they work great with the 
  simple one port forwarded through. It also makes it nice because I use 
idefisk for my softphone on my laptop when I am traveling. Idefisk is a 
small Win32 IAX2 softphone.

http://www.asteriskguru.com/idefisk/

For more Asterisk NAT help see:

http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions

Jonathan