[Asterisk-Users] Asterisk behind NAT to SIP provider

[Rich Adamson]

> > I experimented a little bit and Asterisk behind NAT with SIP works. I created 
> > an account at iptel.org and use that account for outbound SIP traffic from
> > Asterisk.
> I can confirm that Asterisk behind NAT can call out to IPtel.org
> ...and users connected to iptel.org can call me, if my server is registred
> to iptel.org.
> 
> As stated earlier, the iptel.org SIP express router is configured with
> a development version of the nathelper module, that assists SIP clients
> inside a NAT to keep sessions open, allowing incoming calls. In this
> configuration, Asterisk is simply just another SIP phone, seen from
> iptel.org's point of view.
> 
> I'll update the information on the wiki so you can experiment with this.
> 
> Thank you, Jan Janak @iptel.org, for testing with me!

Olle,

That's exactly one of the methods I was referring to in my long-winded
dissertation on asterisk with nat. There are others as well.

It would be nice if some detailed technical explanation was included
in the documentation as to "why" it works, and not just refer to nathelper
as though everyone reading the doc will understand what that module
is actually doing. (It probably won't help the plug-n-play newbies, but
will certainly enlighten those that keep posting unqualified responses
similar to "asterisk won't work behind a nat box".)

If possible, I'd also ensure you test the config with "two" or more
simultanous conversations (through the nat box) as there are likely
to be some limitations that should probably be noted as well.

Rich