Leif, Le dimanche 15 novembre 2009 à 16:18 -0500, Leif Madsen a écrit :
> I'm not sure you've provided enough of the trace here. It finds the peer, but > rejects it with a 401 Unauthorized, which is not uncommon. And I don't see > any > authentication information in the first INVITE. This is why the 401 is sent > back, as the WWW-Authenticate line contains the realm and nonce which should > be > used by the other end to generate the authentication, and then send another > INVITE back with authentication. > > Since you've only shown the two packets in the trace, it is impossible to > tell > what is going on beyond the 401 response from Asterisk. > > Leif. I attach more packets (from a new test run). Thanks, Eric <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '[email protected]' Method: OPTIONS asterisk*CLI> <--- SIP read from UDP:212.27.52.5:5060 ---> INVITE sip:[email protected]:5060;transport=udp SIP/2.0 Call-ID: [email protected] Contact: <sip:172.17.20.241:5062> Content-Type: application/sdp CSeq: 240171363 INVITE From: "Caller" <sip:[email protected];user=phone>;tag=01749-RV-0eb4458b-093bd4160 Max-Forwards: 28 Record-Route: <sip:C=on-88.165.134.117.5060;[email protected]:5060;lr> To: <sip:[email protected];user=phone> Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-XNJD-0079618e-344c12fe Allow: UPDATE,REFER,INFO User-Agent: Cirpack/v4.41c (gw_sip) Content-Length: 173 v=0 o=cp10 125832072268 125832072268 IN IP4 212.27.52.130 s=SIP Call c=IN IP4 212.27.52.130 t=0 0 m=audio 33092 RTP/AVP 8 b=AS:64 a=rtpmap:8 PCMA/8000/1 a=ptime:30 <-------------> --- (13 headers 9 lines) --- == Using SIP RTP CoS mark 5 Sending to 212.27.52.5 : 5060 (no NAT) Using INVITE request as basis request - [email protected] Found peer 'freephonie_appelsortant' for '095245XXXX' from 212.27.52.5:5060 asterisk*CLI> <--- Reliably Transmitting (no NAT) to 212.27.52.5:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-XNJD-0079618e-344c12fe;received=212.27.52.5 From: "Caller" <sip:[email protected];user=phone>;tag=01749-RV-0eb4458b-093bd4160 To: <sip:[email protected];user=phone>;tag=as7249497d Call-ID: [email protected] CSeq: 240171363 INVITE Server: Asterisk PBX 1.6.2.0~rc2-0ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4a3516d7" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE) asterisk*CLI> <--- SIP read from UDP:212.27.52.5:5060 ---> ACK sip:[email protected]:5060;transport=udp SIP/2.0 Call-ID: [email protected] CSeq: 240171363 ACK From: "Caller" <sip:[email protected];user=phone>;tag=01749-RV-0eb4458b-093bd4160 Max-Forwards: 28 To: <sip:[email protected];user=phone>;tag=as7249497d Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-XNJD-0079618e-344c12fe Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '[email protected]' Method: ACK Reliably Transmitting (no NAT) to 212.27.52.5:5060: OPTIONS sip:freephonie.net SIP/2.0 Via: SIP/2.0/UDP 192.168.4.2:5060;branch=z9hG4bK41486e10;rport Max-Forwards: 70 From: "asterisk" <sip:[email protected]>;tag=as096de98b To: <sip:freephonie.net> Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.0~rc2-0ubuntu1 Date: Sun, 15 Nov 2009 21:32:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Reliably Transmitting (no NAT) to 212.27.52.5:5060: OPTIONS sip:freephonie.net SIP/2.0 Via: SIP/2.0/UDP 192.168.4.2:5060;branch=z9hG4bK49b10bb3;rport Max-Forwards: 70 From: "asterisk" <sip:[email protected]>;tag=as0c4787d3 To: <sip:freephonie.net> Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.0~rc2-0ubuntu1 Date: Sun, 15 Nov 2009 21:32:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- asterisk*CLI> <--- SIP read from UDP:212.27.52.5:5060 ---> SIP/2.0 501 Not Implemented Yet Call-ID: [email protected] CSeq: 102 OPTIONS From: "asterisk" <sip:[email protected]>;tag=as096de98b To: <sip:freephonie.net>;tag=00-32733-00efde5e-4f3d2a1b6 Via: SIP/2.0/UDP 192.168.4.2:5060;received=88.165.134.117;rport=5060;branch=z9hG4bK41486e10 Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '[email protected]' Method: OPTIONS asterisk*CLI> <--- SIP read from UDP:212.27.52.5:5060 ---> SIP/2.0 501 Not Implemented Yet Call-ID: [email protected] CSeq: 102 OPTIONS From: "asterisk" <sip:[email protected]>;tag=as0c4787d3 To: <sip:freephonie.net>;tag=00-31015-00efde62-60b13b660 Via: SIP/2.0/UDP 192.168.4.2:5060;received=88.165.134.117;rport=5060;branch=z9hG4bK49b10bb3 Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '[email protected]' Method: OPTIONS Reliably Transmitting (no NAT) to 212.27.52.5:5060: OPTIONS sip:freephonie.net SIP/2.0 Via: SIP/2.0/UDP 192.168.4.2:5060;branch=z9hG4bK2470d659;rport Max-Forwards: 70 From: "asterisk" <sip:[email protected]>;tag=as4b104230 To: <sip:freephonie.net> Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.0~rc2-0ubuntu1 Date: Sun, 15 Nov 2009 21:33:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Reliably Transmitting (no NAT) to 212.27.52.5:5060: OPTIONS sip:freephonie.net SIP/2.0 Via: SIP/2.0/UDP 192.168.4.2:5060;branch=z9hG4bK00867a4e;rport Max-Forwards: 70 From: "asterisk" <sip:[email protected]>;tag=as223230cb To: <sip:freephonie.net> Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.0~rc2-0ubuntu1 Date: Sun, 15 Nov 2009 21:33:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- asterisk*CLI> <--- SIP read from UDP:212.27.52.5:5060 ---> SIP/2.0 501 Not Implemented Yet Call-ID: [email protected] CSeq: 102 OPTIONS From: "asterisk" <sip:[email protected]>;tag=as4b104230 To: <sip:freephonie.net>;tag=00-30943-00efebab-2818f75c0 Via: SIP/2.0/UDP 192.168.4.2:5060;received=88.165.134.117;rport=5060;branch=z9hG4bK2470d659 Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '[email protected]' Method: OPTIONS asterisk*CLI> <--- SIP read from UDP:212.27.52.5:5060 ---> SIP/2.0 501 Not Implemented Yet Call-ID: [email protected] CSeq: 102 OPTIONS From: "asterisk" <sip:[email protected]>;tag=as223230cb To: <sip:freephonie.net>;tag=00-30939-00efebae-7a2b576c6 Via: SIP/2.0/UDP 192.168.4.2:5060;received=88.165.134.117;rport=5060;branch=z9hG4bK00867a4e Content-Length: 0 _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
