Destroying call '77a6e1095722e5f86ef4db404016c8e3@71.41.69.70' asterisk1*CLI> <-- SIP read from 69.90.155.70:5060: INVITE sip:s@71.41.69.70 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 69.90.155.70;branch=z9hG4bKc32e.6e474862.1 Via: SIP/2.0/UDP 192.168.1.114:5060;received=65.35.50.80;branch=z9hG4bK-daeb9f5c From: Derrick Shand ;tag=9f16bda61fe48348o0 To: Call-ID: 9f9dbb7e-5709df50@192.168.1.114 CSeq: 102 INVITE Max-Forwards: 69 Proxy-Authorization: Digest username="765427",realm="fwd.pulver.com",nonce="447e724729bb6037317ad33f53a4efb4d11a97fc",uri="sip:765194@fwd.pulver.com",algorithm=MD5,response="4c0398de6a8488f67eebd3d2c96cef34" Contact: Derrick Shand Expires: 240 User-Agent: Sipura/SPA2000-2.0.11(g) Content-Length: 424 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 680503 680503 IN IP4 192.168.1.114 s=- c=IN IP4 192.168.1.114 t=0 0 m=audio 16452 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (17 headers 19 lines)--- Using INVITE request as basis request - 9f9dbb7e-5709df50@192.168.1.114 Sending to 69.90.155.70 : 5060 (NAT) Found peer 'test' Reliably Transmitting (NAT) to 69.90.155.70:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 69.90.155.70;branch=z9hG4bKc32e.6e474862.1;received=69.90.155.70 Via: SIP/2.0/UDP 192.168.1.114:5060;received=65.35.50.80;branch=z9hG4bK-daeb9f5c From: Derrick Shand ;tag=9f16bda61fe48348o0 To: ;tag=as155b0653 Call-ID: 9f9dbb7e-5709df50@192.168.1.114 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="450c3eeb" Content-Length: 0 --- Scheduling destruction of call '9f9dbb7e-5709df50@192.168.1.114' in 15000 ms asterisk1*CLI> <-- SIP read from 69.90.155.70:5060: ACK sip:s@71.41.69.70 SIP/2.0 Via: SIP/2.0/UDP 69.90.155.70;branch=z9hG4bKc32e.6e474862.1 From: Derrick Shand ;tag=9f16bda61fe48348o0 Call-ID: 9f9dbb7e-5709df50@192.168.1.114 To: ;tag=as155b0653 CSeq: 102 ACK User-Agent: Sip EXpress router(0.8.14-6 (i386/linux)) Content-Length: 0 --- (8 headers 0 lines)--- 12 headers, 0 lines Reliably Transmitting (NAT) to 69.90.155.70:5060: OPTIONS sip:fwd.pulver.com SIP/2.0 Via: SIP/2.0/UDP 71.41.69.70:5060;branch=z9hG4bK026971fc;rport From: "asterisk" ;tag=as7b224b2d To: Contact: Call-ID: 6ee45b3a3467c7101865cd183d064397@71.41.69.70 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 01 Jun 2006 04:39:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- 12 headers, 0 lines Reliably Transmitting (NAT) to 69.90.155.70:5060: OPTIONS sip:fwd.pulver.com SIP/2.0 Via: SIP/2.0/UDP 71.41.69.70:5060;branch=z9hG4bK609c2c1d;rport From: "asterisk" ;tag=as42e66054 To: Contact: Call-ID: 5e6cb6ca3c3579e93d7540897456cb11@71.41.69.70 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 01 Jun 2006 04:39:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- 12 headers, 0 lines Reliably Transmitting (NAT) to 69.90.155.70:5060: OPTIONS sip:fwd.pulver.com SIP/2.0 Via: SIP/2.0/UDP 71.41.69.70:5060;branch=z9hG4bK5117f613;rport From: "asterisk" ;tag=as70d38f70 To: Contact: Call-ID: 4b84a2a73dcc6edc447e79ff2aefffbb@71.41.69.70 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 01 Jun 2006 04:39:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 69.90.155.70:5060: SIP/2.0 404 No such URI here Via: SIP/2.0/UDP 71.41.69.70:5060;branch=z9hG4bK026971fc;rport=5060 From: "asterisk" ;tag=as7b224b2d To: ;tag=cb2000b247d89723001a836145f3b053.2b05 Call-ID: 6ee45b3a3467c7101865cd183d064397@71.41.69.70 CSeq: 102 OPTIONS Server: Sip EXpress router (0.8.14-6 (i386/linux)) Content-Length: 0 --- (8 headers 0 lines)--- Destroying call '6ee45b3a3467c7101865cd183d064397@71.41.69.70' asterisk1*CLI> <-- SIP read from 69.90.155.70:5060: SIP/2.0 404 No such URI here Via: SIP/2.0/UDP 71.41.69.70:5060;branch=z9hG4bK609c2c1d;rport=5060 From: "asterisk" ;tag=as42e66054 To: ;tag=cb2000b247d89723001a836145f3b053.7bcc Call-ID: 5e6cb6ca3c3579e93d7540897456cb11@71.41.69.70 CSeq: 102 OPTIONS Server: Sip EXpress router (0.8.14-6 (i386/linux)) Content-Length: 0 --- (8 headers 0 lines)--- Destroying call '5e6cb6ca3c3579e93d7540897456cb11@71.41.69.70' asterisk1*CLI> <-- SIP read from 69.90.155.70:5060: SIP/2.0 404 No such URI here Via: SIP/2.0/UDP 71.41.69.70:5060;branch=z9hG4bK5117f613;rport=5060 From: "asterisk" ;tag=as70d38f70 To: ;tag=cb2000b247d89723001a836145f3b053.60b9 Call-ID: 4b84a2a73dcc6edc447e79ff2aefffbb@71.41.69.70 CSeq: 102 OPTIONS Server: Sip EXpress router (0.8.14-6 (i386/linux)) Content-Length: 0 --- (8 headers 0 lines)--- Destroying call '4b84a2a73dcc6edc447e79ff2aefffbb@71.41.69.70' 12 headers, 0 lines Reliably Transmitting (NAT) to 65.183.1.18:1262: OPTIONS sip:202@10.10.10.7:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 71.41.69.70:5060;branch=z9hG4bK7cde2765;rport From: "asterisk" ;tag=as2b9e5e49 To: Contact: Call-ID: 7c2fd6aa7294e27139d3f1a6138e24b0@71.41.69.70 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 01 Jun 2006 04:39:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 65.183.1.18:1262: SIP/2.0 200 OK Via: SIP/2.0/UDP 71.41.69.70:5060;branch=z9hG4bK7cde2765;rport From: "asterisk" ;tag=as2b9e5e49 To: ;tag=290228326 Call-ID: 7c2fd6aa7294e27139d3f1a6138e24b0@71.41.69.70 CSeq: 102 OPTIONS Server: Cisco ATA 186 v3.1.0 atasip (040211A) Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER Content-Length: 240 Content-Type: application/sdp v=0 o=202 674187 674187 IN IP4 10.10.10.7 s=ATA186 Call c=IN IP4 10.10.10.7 t=0 0 m=audio 0 RTP/AVP 0 8 4 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:4 G723/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --- (10 headers 11 lines)--- Destroying call '7c2fd6aa7294e27139d3f1a6138e24b0@71.41.69.70'