[Asterisk-Users] Snom 200

[Bill Hamlin]

You must have port mapping in the Linux NAT that allows the SIP-level
packets to get to the * Server, so you need to add a port mapping for the
RTP packets.  I may be wrong but I think * sends RTP on the same port that
it receives RTP on, so once the phone sends some RTP to * then the RTP
coming back should work.

Turn on "sip debug" to see the packets and cut and paste here if you're
still having a problem.

Bill




> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Geert Nijpels
> Sent: Monday, March 22, 2004 4:25 PM
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] Snom 200
>
>
> Barry Fawthrop wrote:
>
> >Progress
> >
> >It seems I can't hear the Say Time, due to RTP Double NAT
> >I'm guess this is both problem 1 and 2 really issue.
> >
> >My config:
> >IP Phone <-> Router (Nat) <-> Internet <-> Linux (NAT) <-> * Server
> >
> >ANyone know of work arounds the double NAT? or other methods
> >to route RTP with snom 200s, to work around this?
> >
> >
> I think you can make progress with the following link:
> http://www.voip-info.org/tiki-index.php?page=NAT%20and%20VOIP
>
> Have fun,
>
> Geert
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>