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: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Geert Nijpels > Sent: Monday, March 22, 2004 4:25 PM > To: [EMAIL PROTECTED] > 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 > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
