[asterisk-users] Asterisk and peers behind nat with port forward, how to proxy?

[Marcus Carlson]

Hi Raul,

Try canreinvite=no in your sip.conf file. Then all calls will go via 
asterisk.

Marcus

Raul Dias skrev:
> Hi,
>
>
> I have the following setup:
>
>
>             [ Voip Provider ] ------ (XX) XXXX-XXXX 
>                  x.x.x.x         (real world phone number)
>                     |
>             { The Internet }
>                     |
>           200.x.x.x (Internet IP)
>              [linux router]
>                 10.0.51.1
>                     |
>         ------------------------- -> (The Lan)
>         |                       |
> [sip peer 1/client]     [asterisk server]
>      10.0.51.3              10.0.51.2
>
>
> The linux router does Nat/firewall for The Lan.
> sip clients inside the Lan can talk to each other (and asterisk) fine.
>
> The router has port forwarding for IAX[2], SIP, RTP (10000-20000) and
> MGCP to the asterisk box (10.0.51.2).
>
> When I have a call between the outside world (VOIP provider) and a
> internal sip peer, I can see that the data transfer (RTP) is between the
> the VOIP provider and the client (10.0.51.3). 
>
>
> That said, the PROBLEM is:
> After a few seconds (2 to 20) the call becomes mute (but still active).
> It does not matter which side started the call.
>
>
> For what I understood, shouldnt asterisk (10.0.51.2) work as a proxy for
> (10.0.51.3), instead of letting it talk directly with the VOIP Provider?
>
> I think that this is where the problem is.  In sip.conf I have externip
> set to the router Internet ip address.  However as the peer is also
> behind the nat (10.0.51.3), the voip provider will see the same IP
> because of nat.  But they are different boxes.
>
>
> - Raul Dias
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>