If I have both clients and Asterisk in the same nat, it is working fine with internal addressing.
When using outside IP with appropriate ports open(5060,10000-2000), with following call flow, X-Lite->asterisk < -- nat-- -> as5400->pstn canreinvite=no nat=yes with this setting RTP's should be between ata186 and astersik over nat. I can see bi-directional RTP streams in Ethereal ( *asterisk<->x-lite* ), very few of them from Asterisk to X-Lite, resulting one-way audio, and the call is disconnected abruptly after that. I have setup g711, on X-Lite and SIP.conf, but still it is negotiating "gsm" with AS5400. Eventually I wan to use clients on different nats, to work with Asterisk on different nat. Is this a codec issue, or asterisk problem or nat? can some body help, probably I need proxy. Obaid Siddiqui. Network Engineer, Prizm Communications, LP Austin, Texas. ----- Original Message ----- From: "Michael J. Tubby G8TIC" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Sent: Sunday, May 29, 2005 7:13 PM Subject: Re: [Asterisk-Users] Peer to Peer calls > > ----- Original Message ----- > From: "Michiel van Baak" <[EMAIL PROTECTED]> > To: <[email protected]> > Sent: Sunday, May 29, 2005 10:41 PM > Subject: Re: [Asterisk-Users] Peer to Peer calls > > > > On 00:32, Mon 30 May 05, Cenk Yabas wrote: > >> Can anybody please answer this. > >> Both clients are behind different NAT's. > >> One of them starts a SIP call to the other through Asterisk. > >> Asterisk sets up the call. > >> Issues reinvite and connects them together. > >> After this point does the media stream flow through Asterisk or Peer to > >> Peer? > >> Does such a call use any system resources of Asterisk server after > >> connection? > >> Thank you in advance. > > > > Did you test this ? > > My experience is the 'reinvite' does not work in the setup > > you descripted. I always have to set 'canreinvite=no' in > > asterisk config or the audio will not come through. > > If you have only one phone on both NAT's and you can do > > port-forwording on both firewalls, it can work, but that > > scenario is highly uncommon. > > The audio stream is setup on some random port, so your > > firewall will block this by default. > > > > *But* If your firewall is SIP-aware - for example a Cisco 837 ADSL > router with IOS 12.3 - then it should be able to fix up the firewall rules > dynamically so that when the phones in the inside (behind the firewall) > re-invite it should inspect the SIP on udp/5060 and see the invitation and > open the appropriate UDP port(s) for the RTP stream. > > Mike > > > > _______________________________________________ > 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
