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<br><br>Don't remember the exact syntax but guess it's something like sip:username@the.phones.ip:5060<br>
<br>Alyed<br><br><br><br><div class="gmail_quote">2010/3/26 haloha <span dir="ltr"><<a href="mailto:haloha201@gmail.com">haloha201@gmail.com</a>></span><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Hi all<br><br>my asterisk server, 2 sip client softphones are the same LAN<br><br>asterisk ip address : 192.168.1.5<br>sip client 1 : 192.168.1.4<br>sip client 2 : 192.168.1.2<br><br>asterisk starts ok with sip<br><br>setup the sip.conf<br>
[test]<br>type=friend<br>username=test<br>secret=1000<br>host=dynamic<br>context=cucku<br>directmedia=yes<br>directrtpsetup=yes<br><br>[1000]<br>type=friend<br>username=1000<br>secret=1000<br>host=dynamic<br>context=cucku<br>
directmedia=yes<br>directrtpsetup=yes<br><br>when make call between 2 sip clients and see the debug in asterisk console<br>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<br>
<br>is there any way to setup rtp directly between 2 sip clients, no need to go through asterisk server<br><br>here is my debug log:<br><--- SIP read from UDP://<a href="http://192.168.1.4:18341" target="_blank">192.168.1.4:18341</a> ---><br>
INVITE <a href="mailto:sip%3A1000@192.168.1.5" target="_blank">sip:1000@192.168.1.5</a> SIP/2.0<br>Via: SIP/2.0/UDP 192.168.1.4:18341;branch=z9hG4bK-d8754z-da695e1f167deb68-1---d8754z-;rport<br>Max-Forwards: 70<br>Contact: <<a href="http://sip:test@192.168.1.4:18341" target="_blank">sip:test@192.168.1.4:18341</a>><br>
To: "1000"<<a href="mailto:sip%3A1000@192.168.1.5" target="_blank">sip:1000@192.168.1.5</a>><br>From: "Do Nguyen Ha"<<a href="mailto:sip%3Atest@192.168.1.5" target="_blank">sip:test@192.168.1.5</a>>;tag=f543a140<br>
Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg.<br>
CSeq: 2 INVITE<br>Content-Type: application/sdp<br>User-Agent: X-Lite release 1104o stamp 56125<br>Content-Length: 261<br>v=0<br>o=- 8 2 IN IP4 192.168.1.4<br>s=CounterPath X-Lite 3.0<br>c=IN IP4 192.168.1.4<br>t=0 0<br>
m=audio 50420 RTP/AVP 107 0 8 101<br>
<br><--- Transmitting (no NAT) to <a href="http://192.168.1.4:18341" target="_blank">192.168.1.4:18341</a> ---><br>SIP/2.0 100 Trying<br>Via: SIP/2.0/UDP 192.168.1.4:18341;branch=z9hG4bK-d8754z-da695e1f167deb68-1---d8754z-;received=192.168.1.4;rport=18341<br>
From: "Do Nguyen Ha"<<a href="mailto:sip%3Atest@192.168.1.5" target="_blank">sip:test@192.168.1.5</a>>;tag=f543a140<br>To: "1000"<<a href="mailto:sip%3A1000@192.168.1.5" target="_blank">sip:1000@192.168.1.5</a>><br>
Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg.<br>
CSeq: 2 INVITE<br>User-Agent: Asterisk PBX 1.6.0.26<br>Supported: replaces, timer<br>Contact: <<a href="mailto:sip%3A1000@192.168.1.5" target="_blank">sip:1000@192.168.1.5</a>><br>Content-Length: 0<br><br>Reliably Transmitting (no NAT) to <a href="http://192.168.1.2:34312" target="_blank">192.168.1.2:34312</a>:<br>
INVITE sip:1000@192.168.1.2:34312;rinstance=862211afcf483176 SIP/2.0<br>Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK2dfa6cce;rport<br>Max-Forwards: 70<br>From: "Do Nguyen Ha" <<a href="mailto:sip%3Atest@192.168.1.5" target="_blank">sip:test@192.168.1.5</a>>;tag=as2886cf30<br>
To: <sip:1000@192.168.1.2:34312;rinstance=862211afcf483176><br>Contact: <<a href="mailto:sip%3Atest@192.168.1.5" target="_blank">sip:test@192.168.1.5</a>><br>Call-ID: <a href="mailto:5fcaa0dc5dd8cac86f01164b2ea6d03a@192.168.1.5" target="_blank">5fcaa0dc5dd8cac86f01164b2ea6d03a@192.168.1.5</a><br>
CSeq: 102 INVITE<br>User-Agent: Asterisk PBX 1.6.0.26<br>Date: Thu, 25 Mar 2010 12:15:05 GMT<br>Supported: replaces, timer<br>Content-Type: application/sdp<br>Content-Length: 309<br>v=0<br>o=root 1983608375 1983608375 IN IP4 192.168.1.5<br>
s=Asterisk PBX 1.6.0.26<br>c=IN IP4 192.168.1.5<br>t=0 0<br>m=audio 17580 RTP/AVP 0 3 8 101<br><br><--- SIP read from UDP://<a href="http://192.168.1.2:34312" target="_blank">192.168.1.2:34312</a> ---><br>SIP/2.0 180 Ringing<br>
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK2dfa6cce;rport=5060<br>
Contact: <sip:1000@192.168.1.2:34312;rinstance=862211afcf483176><br>To: <sip:1000@192.168.1.2:34312;rinstance=862211afcf483176>;tag=403c255c<br>From: "Do Nguyen Ha"<<a href="mailto:sip%3Atest@192.168.1.5" target="_blank">sip:test@192.168.1.5</a>>;tag=as2886cf30<br>
Call-ID: <a href="mailto:5fcaa0dc5dd8cac86f01164b2ea6d03a@192.168.1.5" target="_blank">5fcaa0dc5dd8cac86f01164b2ea6d03a@192.168.1.5</a><br>CSeq: 102 INVITE<br>User-Agent: X-Lite release 1104o stamp 56125<br>Content-Length: 0<br>
<br><br><--- Transmitting (no NAT) to <a href="http://192.168.1.4:18341" target="_blank">192.168.1.4:18341</a> ---><br>
SIP/2.0 180 Ringing<br>Via: SIP/2.0/UDP 192.168.1.4:18341;branch=z9hG4bK-d8754z-da695e1f167deb68-1---d8754z-;received=192.168.1.4;rport=18341<br>From: "Do Nguyen Ha"<<a href="mailto:sip%3Atest@192.168.1.5" target="_blank">sip:test@192.168.1.5</a>>;tag=f543a140<br>
To: "1000"<<a href="mailto:sip%3A1000@192.168.1.5" target="_blank">sip:1000@192.168.1.5</a>>;tag=as0307d0b3<br>Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg.<br>CSeq: 2 INVITE<br>User-Agent: Asterisk PBX 1.6.0.26<br>
Supported: replaces, timer<br>ontact: <<a href="mailto:sip%3A1000@192.168.1.5" target="_blank">sip:1000@192.168.1.5</a>><br>Content-Length: 0<br><br><br><--- SIP read from UDP://<a href="http://192.168.1.2:34312" target="_blank">192.168.1.2:34312</a> ---><br>
SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK2dfa6cce;rport=5060<br>Contact: <sip:1000@192.168.1.2:34312;rinstance=862211afcf483176><br>To: <sip:1000@192.168.1.2:34312;rinstance=862211afcf483176>;tag=403c255c<br>
From: "Do Nguyen Ha"<<a href="mailto:sip%3Atest@192.168.1.5" target="_blank">sip:test@192.168.1.5</a>>;tag=as2886cf30<br>Call-ID: <a href="mailto:5fcaa0dc5dd8cac86f01164b2ea6d03a@192.168.1.5" target="_blank">5fcaa0dc5dd8cac86f01164b2ea6d03a@192.168.1.5</a><br>
CSeq: 102 INVITE<br>Content-Type: application/sdp<br>User-Agent: X-Lite release 1104o stamp 56125<br>Content-Length: 183<br>v=0<br>o=- 8 2 IN IP4 192.168.1.2<br>s=CounterPath X-Lite 3.0<br>c=IN IP4 192.168.1.2<br>t=0 0<br>
m=audio 53062 RTP/AVP 0 8 101<br><br><-------------><br>ACK sip:1000@192.168.1.2:34312;rinstance=862211afcf483176 SIP/2.0<br>Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK2852e1cc;rport<br>Max-Forwards: 70<br>From: "Do Nguyen Ha" <<a href="mailto:sip%3Atest@192.168.1.5" target="_blank">sip:test@192.168.1.5</a>>;tag=as2886cf30<br>
To: <sip:1000@192.168.1.2:34312;rinstance=862211afcf483176>;tag=403c255c<br>Contact: <<a href="mailto:sip%3Atest@192.168.1.5" target="_blank">sip:test@192.168.1.5</a>><br>Call-ID: <a href="mailto:5fcaa0dc5dd8cac86f01164b2ea6d03a@192.168.1.5" target="_blank">5fcaa0dc5dd8cac86f01164b2ea6d03a@192.168.1.5</a><br>
CSeq: 102 ACK<br>User-Agent: Asterisk PBX 1.6.0.26<br>Content-Length: 0<br><br><--- Reliably Transmitting (no NAT) to <a href="http://192.168.1.4:18341" target="_blank">192.168.1.4:18341</a> ---><br>SIP/2.0 200 OK<br>
Via: SIP/2.0/UDP 192.168.1.4:18341;branch=z9hG4bK-d8754z-da695e1f167deb68-1---d8754z-;received=192.168.1.4;rport=18341<br>
From: "Do Nguyen Ha"<<a href="mailto:sip%3Atest@192.168.1.5" target="_blank">sip:test@192.168.1.5</a>>;tag=f543a140<br>To: "1000"<<a href="mailto:sip%3A1000@192.168.1.5" target="_blank">sip:1000@192.168.1.5</a>>;tag=as0307d0b3<br>
Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg.<br>CSeq: 2 INVITE<br>User-Agent: Asterisk PBX 1.6.0.26<br>Supported: replaces, timer<br>Contact: <<a href="mailto:sip%3A1000@192.168.1.5" target="_blank">sip:1000@192.168.1.5</a>><br>
Content-Type: application/sdp<br>Content-Length: 286<br>v=0<br>o=root 1290114102 1290114102 IN IP4 192.168.1.5<br>s=Asterisk PBX 1.6.0.26<br>c=IN IP4 192.168.1.5<br>t=0 0<br>m=audio 18366 RTP/AVP 0 8 101<br><br><br>Reliably Transmitting (no NAT) to <a href="http://192.168.1.2:34312" target="_blank">192.168.1.2:34312</a>:<br>
INVITE sip:1000@192.168.1.2:34312;rinstance=862211afcf483176 SIP/2.0<br>Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK39e834a9;rport<br>Max-Forwards: 70<br>From: "Do Nguyen Ha" <<a href="mailto:sip%3Atest@192.168.1.5" target="_blank">sip:test@192.168.1.5</a>>;tag=as2886cf30<br>
To: <sip:1000@192.168.1.2:34312;rinstance=862211afcf483176>;tag=403c255c<br>Contact: <<a href="mailto:sip%3Atest@192.168.1.5" target="_blank">sip:test@192.168.1.5</a>><br>Call-ID: <a href="mailto:5fcaa0dc5dd8cac86f01164b2ea6d03a@192.168.1.5" target="_blank">5fcaa0dc5dd8cac86f01164b2ea6d03a@192.168.1.5</a><br>
CSeq: 103 INVITE<br>User-Agent: Asterisk PBX 1.6.0.26<br>Supported: replaces, timer<br>X-asterisk-Info: SIP re-invite (External RTP bridge)<br>Content-Type: application/sdp<br>Content-Length: 286<br>v=0<br>o=root 1983608375 1983608376 IN IP4 192.168.1.4<br>
s=Asterisk PBX 1.6.0.26<br>c=IN IP4 192.168.1.4<br>t=0 0<br>m=audio 50420 RTP/AVP 0 8 101<br><br><--- SIP read from UDP://<a href="http://192.168.1.2:34312" target="_blank">192.168.1.2:34312</a> ---><br>SIP/2.0 100 Trying<br>
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK39e834a9;rport=5060<br>
To: <sip:1000@192.168.1.2:34312;rinstance=862211afcf483176>;tag=403c255c<br>From: "Do Nguyen Ha" <<a href="mailto:sip%3Atest@192.168.1.5" target="_blank">sip:test@192.168.1.5</a>>;tag=as2886cf30<br>
Call-ID: <a href="mailto:5fcaa0dc5dd8cac86f01164b2ea6d03a@192.168.1.5" target="_blank">5fcaa0dc5dd8cac86f01164b2ea6d03a@192.168.1.5</a><br>
CSeq: 103 INVITE<br>Content-Length: 0<br><br><br><--- SIP read from UDP://<a href="http://192.168.1.4:18341" target="_blank">192.168.1.4:18341</a> ---><br>ACK <a href="mailto:sip%3A1000@192.168.1.5" target="_blank">sip:1000@192.168.1.5</a> SIP/2.0<br>
Via: SIP/2.0/UDP 192.168.1.4:18341;branch=z9hG4bK-d8754z-e104ab75c9163459-1---d8754z-;rport<br>Max-Forwards: 70<br>Contact: <<a href="http://sip:test@192.168.1.4:18341" target="_blank">sip:test@192.168.1.4:18341</a>><br>
To: "1000"<<a href="mailto:sip%3A1000@192.168.1.5" target="_blank">sip:1000@192.168.1.5</a>>;tag=as0307d0b3<br>
From: "Do Nguyen Ha"<<a href="mailto:sip%3Atest@192.168.1.5" target="_blank">sip:test@192.168.1.5</a>>;tag=f543a140<br>Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg.<br>CSeq: 2 ACK<br>User-Agent: X-Lite release 1104o stamp 56125<br>
Authorization: Digest username="test",realm="asterisk",nonce="44b4dd5e",uri="<a href="mailto:sip%3A1000@192.168.1.5" target="_blank">sip:1000@192.168.1.5</a>",response="540173a06f742b7f11cde8010f90ec26",algorithm=MD5<br>
Content-Length: 0<br><br><br><-------------><br>Reliably Transmitting (no NAT) to <a href="http://192.168.1.4:18341" target="_blank">192.168.1.4:18341</a>:<br>INVITE <a href="http://sip:test@192.168.1.4:18341" target="_blank">sip:test@192.168.1.4:18341</a> SIP/2.0<br>
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK584d3aaf;rport<br>Max-Forwards: 70<br>From: "1000"<<a href="mailto:sip%3A1000@192.168.1.5" target="_blank">sip:1000@192.168.1.5</a>>;tag=as0307d0b3<br>To: "Do Nguyen Ha"<<a href="mailto:sip%3Atest@192.168.1.5" target="_blank">sip:test@192.168.1.5</a>>;tag=f543a140<br>
Contact: <<a href="mailto:sip%3A1000@192.168.1.5" target="_blank">sip:1000@192.168.1.5</a>><br>Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg.<br>CSeq: 102 INVITE<br>User-Agent: Asterisk PBX 1.6.0.26<br>Supported: replaces, timer<br>
X-asterisk-Info: SIP re-invite (External RTP bridge)<br>Content-Type: application/sdp<br>Content-Length: 286<br>v=0<br>o=root 1290114102 1290114103 IN IP4 192.168.1.2<br>s=Asterisk PBX 1.6.0.26<br>c=IN IP4 192.168.1.2<br>
t=0 0<br>m=audio 53062 RTP/AVP 0 8 101<br><br><--- SIP read from UDP://<a href="http://192.168.1.4:18341" target="_blank">192.168.1.4:18341</a> ---><br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK584d3aaf;rport=5060<br>
Contact: <<a href="http://sip:test@192.168.1.4:18341" target="_blank">sip:test@192.168.1.4:18341</a>><br>To: "Do Nguyen Ha"<<a href="mailto:sip%3Atest@192.168.1.5" target="_blank">sip:test@192.168.1.5</a>>;tag=f543a140<br>
From: "1000"<<a href="mailto:sip%3A1000@192.168.1.5" target="_blank">sip:1000@192.168.1.5</a>>;tag=as0307d0b3<br>
Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg.<br>CSeq: 102 INVITE<br>Content-Type: application/sdp<br>User-Agent: X-Lite release 1104o stamp 56125<br>Content-Length: 183<br>v=0<br>o=- 8 3 IN IP4 192.168.1.4<br>s=CounterPath X-Lite 3.0<br>
c=IN IP4 192.168.1.4<br>t=0 0<br>m=audio 50420 RTP/AVP 0 8 101<br><br><-------------><br>Transmitting (no NAT) to <a href="http://192.168.1.4:18341" target="_blank">192.168.1.4:18341</a>:<br>ACK <a href="http://sip:test@192.168.1.4:18341" target="_blank">sip:test@192.168.1.4:18341</a> SIP/2.0<br>
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK5dde1d6e;rport<br>Max-Forwards: 70<br>From: "1000"<<a href="mailto:sip%3A1000@192.168.1.5" target="_blank">sip:1000@192.168.1.5</a>>;tag=as0307d0b3<br>To: "Do Nguyen Ha"<<a href="mailto:sip%3Atest@192.168.1.5" target="_blank">sip:test@192.168.1.5</a>>;tag=f543a140<br>
Contact: <<a href="mailto:sip%3A1000@192.168.1.5" target="_blank">sip:1000@192.168.1.5</a>><br>Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg.<br>CSeq: 102 ACK<br>User-Agent: Asterisk PBX 1.6.0.26<br>Content-Length: 0<br>
<br>
<--- SIP read from UDP://<a href="http://192.168.1.2:34312" target="_blank">192.168.1.2:34312</a> ---><br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK39e834a9;rport=5060<br>Contact: <sip:1000@192.168.1.2:34312;rinstance=862211afcf483176><br>
To: <sip:1000@192.168.1.2:34312;rinstance=862211afcf483176>;tag=403c255c<br>From: "Do Nguyen Ha"<<a href="mailto:sip%3Atest@192.168.1.5" target="_blank">sip:test@192.168.1.5</a>>;tag=as2886cf30<br>Call-ID: <a href="mailto:5fcaa0dc5dd8cac86f01164b2ea6d03a@192.168.1.5" target="_blank">5fcaa0dc5dd8cac86f01164b2ea6d03a@192.168.1.5</a><br>
CSeq: 103 INVITE<br>Content-Type: application/sdp<br>User-Agent: X-Lite release 1104o stamp 56125<br>Content-Length: 183<br>v=0<br>o=- 8 2 IN IP4 192.168.1.2<br>s=CounterPath X-Lite 3.0<br>c=IN IP4 192.168.1.2<br>t=0 0<br>
m=audio 53062 RTP/AVP 0 8 101<br><br>Thank you<br><br>
<br>--<br>
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