I guess to do what you want you need to dial directly between the phones. Can't do it with xlite but you can with SJphones
Don't remember the exact syntax but guess it's something like sip:[email protected]:5060 Alyed 2010/3/26 haloha <[email protected]> > Hi all > > my asterisk server, 2 sip client softphones are the same LAN > > asterisk ip address : 192.168.1.5 > sip client 1 : 192.168.1.4 > sip client 2 : 192.168.1.2 > > asterisk starts ok with sip > > setup the sip.conf > [test] > type=friend > username=test > secret=1000 > host=dynamic > context=cucku > directmedia=yes > directrtpsetup=yes > > [1000] > type=friend > username=1000 > secret=1000 > host=dynamic > context=cucku > directmedia=yes > directrtpsetup=yes > > when make call between 2 sip clients and see the debug in asterisk console > the problem is asterisk setup the inital call for media = asterisk IP > address, when all things done, asterisk does re-invite to setup the rtp > directly between 2 sip clients > > is there any way to setup rtp directly between 2 sip clients, no need to go > through asterisk server > > here is my debug log: > <--- SIP read from UDP://192.168.1.4:18341 ---> > INVITE sip:[email protected] <sip%[email protected]> SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.4:18341 > ;branch=z9hG4bK-d8754z-da695e1f167deb68-1---d8754z-;rport > Max-Forwards: 70 > Contact: <sip:[email protected]:18341> > To: "1000"<sip:[email protected] <sip%[email protected]>> > From: "Do Nguyen Ha"<sip:[email protected] <sip%[email protected]> > >;tag=f543a140 > Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg. > CSeq: 2 INVITE > Content-Type: application/sdp > User-Agent: X-Lite release 1104o stamp 56125 > Content-Length: 261 > v=0 > o=- 8 2 IN IP4 192.168.1.4 > s=CounterPath X-Lite 3.0 > c=IN IP4 192.168.1.4 > t=0 0 > m=audio 50420 RTP/AVP 107 0 8 101 > > <--- Transmitting (no NAT) to 192.168.1.4:18341 ---> > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 192.168.1.4:18341 > ;branch=z9hG4bK-d8754z-da695e1f167deb68-1---d8754z-;received=192.168.1.4;rport=18341 > From: "Do Nguyen Ha"<sip:[email protected] <sip%[email protected]> > >;tag=f543a140 > To: "1000"<sip:[email protected] <sip%[email protected]>> > Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg. > CSeq: 2 INVITE > User-Agent: Asterisk PBX 1.6.0.26 > Supported: replaces, timer > Contact: <sip:[email protected] <sip%[email protected]>> > Content-Length: 0 > > Reliably Transmitting (no NAT) to 192.168.1.2:34312: > INVITE sip:[email protected]:34312;rinstance=862211afcf483176 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK2dfa6cce;rport > Max-Forwards: 70 > From: "Do Nguyen Ha" <sip:[email protected] <sip%[email protected]> > >;tag=as2886cf30 > To: <sip:[email protected]:34312;rinstance=862211afcf483176> > Contact: <sip:[email protected] <sip%[email protected]>> > Call-ID: [email protected] > CSeq: 102 INVITE > User-Agent: Asterisk PBX 1.6.0.26 > Date: Thu, 25 Mar 2010 12:15:05 GMT > Supported: replaces, timer > Content-Type: application/sdp > Content-Length: 309 > v=0 > o=root 1983608375 1983608375 IN IP4 192.168.1.5 > s=Asterisk PBX 1.6.0.26 > c=IN IP4 192.168.1.5 > t=0 0 > m=audio 17580 RTP/AVP 0 3 8 101 > > <--- SIP read from UDP://192.168.1.2:34312 ---> > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK2dfa6cce;rport=5060 > Contact: <sip:[email protected]:34312;rinstance=862211afcf483176> > To: <sip:[email protected]:34312;rinstance=862211afcf483176>;tag=403c255c > From: "Do Nguyen Ha"<sip:[email protected] <sip%[email protected]> > >;tag=as2886cf30 > Call-ID: [email protected] > CSeq: 102 INVITE > User-Agent: X-Lite release 1104o stamp 56125 > Content-Length: 0 > > > <--- Transmitting (no NAT) to 192.168.1.4:18341 ---> > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 192.168.1.4:18341 > ;branch=z9hG4bK-d8754z-da695e1f167deb68-1---d8754z-;received=192.168.1.4;rport=18341 > From: "Do Nguyen Ha"<sip:[email protected] <sip%[email protected]> > >;tag=f543a140 > To: "1000"<sip:[email protected] <sip%[email protected]>>;tag=as0307d0b3 > Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg. > CSeq: 2 INVITE > User-Agent: Asterisk PBX 1.6.0.26 > Supported: replaces, timer > ontact: <sip:[email protected] <sip%[email protected]>> > Content-Length: 0 > > > <--- SIP read from UDP://192.168.1.2:34312 ---> > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK2dfa6cce;rport=5060 > Contact: <sip:[email protected]:34312;rinstance=862211afcf483176> > To: <sip:[email protected]:34312;rinstance=862211afcf483176>;tag=403c255c > From: "Do Nguyen Ha"<sip:[email protected] <sip%[email protected]> > >;tag=as2886cf30 > Call-ID: [email protected] > CSeq: 102 INVITE > Content-Type: application/sdp > User-Agent: X-Lite release 1104o stamp 56125 > Content-Length: 183 > v=0 > o=- 8 2 IN IP4 192.168.1.2 > s=CounterPath X-Lite 3.0 > c=IN IP4 192.168.1.2 > t=0 0 > m=audio 53062 RTP/AVP 0 8 101 > > <-------------> > ACK sip:[email protected]:34312;rinstance=862211afcf483176 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK2852e1cc;rport > Max-Forwards: 70 > From: "Do Nguyen Ha" <sip:[email protected] <sip%[email protected]> > >;tag=as2886cf30 > To: <sip:[email protected]:34312;rinstance=862211afcf483176>;tag=403c255c > Contact: <sip:[email protected] <sip%[email protected]>> > Call-ID: [email protected] > CSeq: 102 ACK > User-Agent: Asterisk PBX 1.6.0.26 > Content-Length: 0 > > <--- Reliably Transmitting (no NAT) to 192.168.1.4:18341 ---> > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.1.4:18341 > ;branch=z9hG4bK-d8754z-da695e1f167deb68-1---d8754z-;received=192.168.1.4;rport=18341 > From: "Do Nguyen Ha"<sip:[email protected] <sip%[email protected]> > >;tag=f543a140 > To: "1000"<sip:[email protected] <sip%[email protected]>>;tag=as0307d0b3 > Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg. > CSeq: 2 INVITE > User-Agent: Asterisk PBX 1.6.0.26 > Supported: replaces, timer > Contact: <sip:[email protected] <sip%[email protected]>> > Content-Type: application/sdp > Content-Length: 286 > v=0 > o=root 1290114102 1290114102 IN IP4 192.168.1.5 > s=Asterisk PBX 1.6.0.26 > c=IN IP4 192.168.1.5 > t=0 0 > m=audio 18366 RTP/AVP 0 8 101 > > > Reliably Transmitting (no NAT) to 192.168.1.2:34312: > INVITE sip:[email protected]:34312;rinstance=862211afcf483176 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK39e834a9;rport > Max-Forwards: 70 > From: "Do Nguyen Ha" <sip:[email protected] <sip%[email protected]> > >;tag=as2886cf30 > To: <sip:[email protected]:34312;rinstance=862211afcf483176>;tag=403c255c > Contact: <sip:[email protected] <sip%[email protected]>> > Call-ID: [email protected] > CSeq: 103 INVITE > User-Agent: Asterisk PBX 1.6.0.26 > Supported: replaces, timer > X-asterisk-Info: SIP re-invite (External RTP bridge) > Content-Type: application/sdp > Content-Length: 286 > v=0 > o=root 1983608375 1983608376 IN IP4 192.168.1.4 > s=Asterisk PBX 1.6.0.26 > c=IN IP4 192.168.1.4 > t=0 0 > m=audio 50420 RTP/AVP 0 8 101 > > <--- SIP read from UDP://192.168.1.2:34312 ---> > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK39e834a9;rport=5060 > To: <sip:[email protected]:34312;rinstance=862211afcf483176>;tag=403c255c > From: "Do Nguyen Ha" <sip:[email protected] <sip%[email protected]> > >;tag=as2886cf30 > Call-ID: [email protected] > CSeq: 103 INVITE > Content-Length: 0 > > > <--- SIP read from UDP://192.168.1.4:18341 ---> > ACK sip:[email protected] <sip%[email protected]> SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.4:18341 > ;branch=z9hG4bK-d8754z-e104ab75c9163459-1---d8754z-;rport > Max-Forwards: 70 > Contact: <sip:[email protected]:18341> > To: "1000"<sip:[email protected] <sip%[email protected]>>;tag=as0307d0b3 > From: "Do Nguyen Ha"<sip:[email protected] <sip%[email protected]> > >;tag=f543a140 > Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg. > CSeq: 2 ACK > User-Agent: X-Lite release 1104o stamp 56125 > Authorization: Digest > username="test",realm="asterisk",nonce="44b4dd5e",uri=" > sip:[email protected] <sip%[email protected]> > ",response="540173a06f742b7f11cde8010f90ec26",algorithm=MD5 > Content-Length: 0 > > > <-------------> > Reliably Transmitting (no NAT) to 192.168.1.4:18341: > INVITE sip:[email protected]:18341 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK584d3aaf;rport > Max-Forwards: 70 > From: "1000"<sip:[email protected] <sip%[email protected]>>;tag=as0307d0b3 > To: "Do Nguyen Ha"<sip:[email protected] <sip%[email protected]> > >;tag=f543a140 > Contact: <sip:[email protected] <sip%[email protected]>> > Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg. > CSeq: 102 INVITE > User-Agent: Asterisk PBX 1.6.0.26 > Supported: replaces, timer > X-asterisk-Info: SIP re-invite (External RTP bridge) > Content-Type: application/sdp > Content-Length: 286 > v=0 > o=root 1290114102 1290114103 IN IP4 192.168.1.2 > s=Asterisk PBX 1.6.0.26 > c=IN IP4 192.168.1.2 > t=0 0 > m=audio 53062 RTP/AVP 0 8 101 > > <--- SIP read from UDP://192.168.1.4:18341 ---> > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK584d3aaf;rport=5060 > Contact: <sip:[email protected]:18341> > To: "Do Nguyen Ha"<sip:[email protected] <sip%[email protected]> > >;tag=f543a140 > From: "1000"<sip:[email protected] <sip%[email protected]>>;tag=as0307d0b3 > Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg. > CSeq: 102 INVITE > Content-Type: application/sdp > User-Agent: X-Lite release 1104o stamp 56125 > Content-Length: 183 > v=0 > o=- 8 3 IN IP4 192.168.1.4 > s=CounterPath X-Lite 3.0 > c=IN IP4 192.168.1.4 > t=0 0 > m=audio 50420 RTP/AVP 0 8 101 > > <-------------> > Transmitting (no NAT) to 192.168.1.4:18341: > ACK sip:[email protected]:18341 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK5dde1d6e;rport > Max-Forwards: 70 > From: "1000"<sip:[email protected] <sip%[email protected]>>;tag=as0307d0b3 > To: "Do Nguyen Ha"<sip:[email protected] <sip%[email protected]> > >;tag=f543a140 > Contact: <sip:[email protected] <sip%[email protected]>> > Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg. > CSeq: 102 ACK > User-Agent: Asterisk PBX 1.6.0.26 > Content-Length: 0 > > <--- SIP read from UDP://192.168.1.2:34312 ---> > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK39e834a9;rport=5060 > Contact: <sip:[email protected]:34312;rinstance=862211afcf483176> > To: <sip:[email protected]:34312;rinstance=862211afcf483176>;tag=403c255c > From: "Do Nguyen Ha"<sip:[email protected] <sip%[email protected]> > >;tag=as2886cf30 > Call-ID: [email protected] > CSeq: 103 INVITE > Content-Type: application/sdp > User-Agent: X-Lite release 1104o stamp 56125 > Content-Length: 183 > v=0 > o=- 8 2 IN IP4 192.168.1.2 > s=CounterPath X-Lite 3.0 > c=IN IP4 192.168.1.2 > t=0 0 > m=audio 53062 RTP/AVP 0 8 101 > > Thank you > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
