[Asterisk-Users] SIP & IAX behind NAT

[Alejandro Ruiz]

I am trying to achieve the same thing.
I have bothe asterisk and X-lite behind NAT.

sip uses port 5060
X-lite can be configured to use an rtp port, and you can specify your
external address...
y configured my nat to foward 5060 to my *
ports 8000 and 8001 too.
I also tried with the sip with no nat. and my result where the same...
I have 5 seconds of audio... so I guess there is another extra port involved
in the operation, one which signals the communication status.
does someone knows anything about this? is my guessing correct?
thanks,
alito


----- Original Message -----
From: Peter Zeltins <peter at fintrading.com>
To: Asterisk Users <asterisk-users at lists.digium.com>
Sent: Monday, October 27, 2003 10:42 AM
Subject: Re: [Asterisk-Users] SIP & IAX behind NAT


> OK, I got IAX to work. Which SIP bone I should break to force it into
> working? ;)
>
> Peter
>
> > IAX should work well behind NAT without further configuration than just
> > a single port forward. RTP based protocols such as MGCP, SIP and H.323
> > require helper agents on the NAT box to work, although SIP can be forced
> > to work by slightly breaking it.
> >
> > roy
> >
> > On Mon, 2003-10-27 at 10:00, Peter Zeltins wrote:
> > > I'm trying to set up * server behind NAT. The box is set up as DMZ in
> > > my DSL router, i.e. all incoming connections without explicit port
> > > mapping are forwarded to *. So far I'm unable to get this setup to
> > > work for either IAX or SIP (tried IAXComm & XLite softphones on public
> > > IP address). Data seems to come in fine (IAX/SIP debug shows message
> > > interaction taking place), but there is no audio.
> > >
> > > Am I missing something in */router setup or such thing cannot be done?
> > > I suspect that I need to override local IP address being sent out in
> > > SIP/IAX packets, but I see no such option in either iax.conf or
> > > sip.conf
> > >
> > > TIA,
> > > Peter
> >
> >
>
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