From: "23002" ;tag=aabpsKuK5JPcWtSE To: "23002" Call-ID: owiz95Hz9XJxvP8b@89.35.79.94 CSeq: 12407 REGISTER Contact: Expires: 60 Authorization: Digest username="23002", realm="asterisk", nonce="3b055b23", uri="sip:voip.telefonet.ro", response="e40b04325817245090db2d9d919f84c8" Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 89.35.79.94 : 5060 (no NAT) <--- Transmitting (no NAT) to 89.35.79.94:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 89.35.79.94:5060;branch=z9hG4bKIpZoJszot;received=89.35.79.94 From: "23002" ;tag=aabpsKuK5JPcWtSE To: "23002" Call-ID: owiz95Hz9XJxvP8b@89.35.79.94 CSeq: 12407 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> voip*CLI> <--- Transmitting (no NAT) to 89.35.79.94:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 89.35.79.94:5060;branch=z9hG4bKIpZoJszot;received=89.35.79.94 From: "23002" ;tag=aabpsKuK5JPcWtSE To: "23002" ;tag=as26ffa75b Call-ID: owiz95Hz9XJxvP8b@89.35.79.94 CSeq: 12407 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Mon, 05 Mar 2007 02:27:24 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'owiz95Hz9XJxvP8b@89.35.79.94' in 32000 ms (Method: REGISTER) voip*CLI> Scheduling destruction of SIP dialog '1970aa3b1b4b2cb87544201012352ae2@89.35.79.5' in 32000 ms (Method: NOTIFY) Reliably Transmitting (no NAT) to 89.35.79.247:5060: NOTIFY sip:23001@89.35.79.247:5060 SIP/2.0 Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK1be45660;rport From: "asterisk" ;tag=as48689ba2 To: Contact: Call-ID: 1970aa3b1b4b2cb87544201012352ae2@89.35.79.5 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 91 Messages-Waiting: yes Message-Account: sip:asterisk@89.35.79.5 Voice-Message: 1/0 (0/0) --- voip*CLI> <--- SIP read from 89.35.79.247:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK1be45660;rport Call-ID: 1970aa3b1b4b2cb87544201012352ae2@89.35.79.5 CSeq: 102 NOTIFY From: "asterisk" ;tag=as48689ba2 To: ;tag=HXL4XOOmb9oG7f9l Contact: Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '1970aa3b1b4b2cb87544201012352ae2@89.35.79.5' Method: NOTIFY Scheduling destruction of SIP dialog '588ae2d34868bd283eeb09b42cc02107@89.35.79.5' in 32000 ms (Method: NOTIFY) Reliably Transmitting (no NAT) to 89.35.79.94:5060: NOTIFY sip:23002@89.35.79.94:5060 SIP/2.0 Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK13d2c8e6;rport From: "asterisk" ;tag=as3c1621ae To: Contact: Call-ID: 588ae2d34868bd283eeb09b42cc02107@89.35.79.5 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 90 Messages-Waiting: no Message-Account: sip:asterisk@89.35.79.5 Voice-Message: 0/0 (0/0) --- voip*CLI> <--- SIP read from 89.35.79.94:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK13d2c8e6;rport Call-ID: 588ae2d34868bd283eeb09b42cc02107@89.35.79.5 CSeq: 102 NOTIFY From: "asterisk" ;tag=as3c1621ae To: ;tag=Hs3pN1qXMhUlF7VQ Contact: Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '588ae2d34868bd283eeb09b42cc02107@89.35.79.5' Method: NOTIFY voip*CLI> <--- SIP read from 89.35.79.247:5060 ---> INVITE sip:23003@voip.telefonet.ro SIP/2.0 Via: SIP/2.0/UDP 89.35.79.247:5060;branch=z9hG4bKW227kVZvZ Max-Forwards: 70 User-Agent: Wu Chuan From: "23001" ;tag=YG6UP1oYxeS4ilVp To: "23003" Call-ID: SPbbitaAMsMutV8Q@89.35.79.247 Contact: CSeq: 1 INVITE Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, REFER, NOTIFY, INFO, PRACK, UPDATE Supported: 100rel, replaces Content-Type: application/sdp Content-Length: 215 v=0 o=23001 23013746 72060732 IN IP4 89.35.79.247 s=SIP CALL c=IN IP4 89.35.79.247 t=0 0 m=audio 30000 RTP/AVP 18 4 0 8 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 <-------------> --- (13 headers 10 lines) --- Sending to 89.35.79.247 : 5060 (no NAT) Using INVITE request as basis request - SPbbitaAMsMutV8Q@89.35.79.247 Found peer '23001' <--- Reliably Transmitting (no NAT) to 89.35.79.247:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 89.35.79.247:5060;branch=z9hG4bKW227kVZvZ;received=89.35.79.247 From: "23001" ;tag=YG6UP1oYxeS4ilVp To: "23003" ;tag=as5ad4b123 Call-ID: SPbbitaAMsMutV8Q@89.35.79.247 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="25942128" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'SPbbitaAMsMutV8Q@89.35.79.247' in 32000 ms (Method: INVITE) voip*CLI> <--- SIP read from 89.35.79.247:5060 ---> ACK sip:23003@voip.telefonet.ro SIP/2.0 Via: SIP/2.0/UDP 89.35.79.247:5060;branch=z9hG4bKW227kVZvZ Max-Forwards: 70 User-Agent: Wu Chuan From: "23001" ;tag=YG6UP1oYxeS4ilVp To: "23003" ;tag=as5ad4b123 Call-ID: SPbbitaAMsMutV8Q@89.35.79.247 Contact: CSeq: 1 ACK Content-Length: 0 <-------------> --- (10 headers 0 lines) --- voip*CLI> <--- SIP read from 89.35.79.247:5060 ---> INVITE sip:23003@voip.telefonet.ro SIP/2.0 Via: SIP/2.0/UDP 89.35.79.247:5060;branch=z9hG4bKfS4gesXYv Max-Forwards: 70 User-Agent: Wu Chuan From: "23001" ;tag=YG6UP1oYxeS4ilVp To: "23003" Call-ID: SPbbitaAMsMutV8Q@89.35.79.247 Contact: Proxy-Authorization: Digest username="23001", realm="asterisk", nonce="25942128", uri="sip:23003@voip.telefonet.ro", response="b12da7f6a02fc62a45fe9453d928853d" CSeq: 2 INVITE Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, REFER, NOTIFY, INFO, PRACK, UPDATE Supported: 100rel, replaces Content-Type: application/sdp Content-Length: 215 v=0 o=23001 67637671 39620179 IN IP4 89.35.79.247 s=SIP CALL c=IN IP4 89.35.79.247 t=0 0 m=audio 30000 RTP/AVP 18 4 0 8 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 <-------------> --- (14 headers 10 lines) --- Sending to 89.35.79.247 : 5060 (no NAT) Using INVITE request as basis request - SPbbitaAMsMutV8Q@89.35.79.247 Found peer '23001' Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 0 Found RTP audio format 8 Peer audio RTP is at port 89.35.79.247:30000 Found description format G729 for ID 18 Found description format G723 for ID 4 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 89.35.79.247:30000 Looking for 23003 in numberplan-custom-1 (domain voip.telefonet.ro) list_route: hop: <--- Transmitting (no NAT) to 89.35.79.247:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 89.35.79.247:5060;branch=z9hG4bKfS4gesXYv;received=89.35.79.247 From: "23001" ;tag=YG6UP1oYxeS4ilVp To: "23003" Call-ID: SPbbitaAMsMutV8Q@89.35.79.247 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [23003@numberplan-custom-1:1] Macro("SIP/23001-b5b0f3d8", "trunkdial|SIP/trunk_1/23003") in new stack -- Executing [s@macro-trunkdial:1] Dial("SIP/23001-b5b0f3d8", "SIP/trunk_1/23003") in new stack Audio is at 10.0.0.2 port 19902 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.0.0.1:5060: INVITE sip:23003@10.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK05b1673f;rport From: "Test" ;tag=as67d70ee4 To: Contact: Call-ID: 47b5de4a4d5113581f9bc6c81d2becb7@10.0.0.2 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 05 Mar 2007 02:27:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 262 v=0 o=root 6308 6308 IN IP4 89.35.79.247 s=session c=IN IP4 89.35.79.247 t=0 0 m=audio 30000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called trunk_1/23003 voip*CLI> <--- SIP read from 10.0.0.1:5060 ---> SIP/2.0 100 Trying From: "Test";tag=as67d70ee4 To: ;tag=a000001-13c4-96f-24f9c8-a0f Call-ID: 47b5de4a4d5113581f9bc6c81d2becb7@10.0.0.2 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.0.0.2:5060;received=10.0.0.2;branch=z9hG4bK05b1673f;rport Contact: Content-Length:0 <-------------> --- (8 headers 0 lines) --- voip*CLI> <--- SIP read from 10.0.0.1:5060 ---> INVITE sip:23003@10.0.0.2;transport=UDP;TCID-27 SIP/2.0 From: "TEST";tag=a000001-13c4-970-24fb54-382b To: Call-ID: 9794353c-a000001-13c4-970-24fb54-3ca5@10.0.0.1 CSeq: 1 INVITE Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-970-24fb54-3488 Max-Forwards: 70 Contact: Content-Type: application/SDP Content-Length:288 v=0 o=a0027 1234567890 1234567890 IN IP4 10.0.0.1 s=SIP Call test c=IN IP4 10.0.0.1 t=0 0 m=audio 10000 RTP/AVP 8 97 0 18 a=fmtp:97 0-15 a=fmtp:18 annexb=yes a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 <-------------> --- (10 headers 13 lines) --- Sending to 10.0.0.1 : 5060 (no NAT) Using INVITE request as basis request - 9794353c-a000001-13c4-970-24fb54-3ca5@10.0.0.1 Found peer 'trunk_1' Found RTP audio format 8 Found RTP audio format 97 Found RTP audio format 0 Found RTP audio format 18 Peer audio RTP is at port 10.0.0.1:10000 Found description format PCMA for ID 8 Found description format telephone-event for ID 97 Found description format PCMU for ID 0 Found description format G729 for ID 18 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.0.0.1:10000 Looking for 23003 in DID_trunk_1 (domain 10.0.0.2) list_route: hop: voip*CLI> <--- Transmitting (no NAT) to 10.0.0.1:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-970-24fb54-3488;received=10.0.0.1 From: "TEST";tag=a000001-13c4-970-24fb54-382b To: Call-ID: 9794353c-a000001-13c4-970-24fb54-3ca5@10.0.0.1 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [23003@DID_trunk_1:1] Dial("SIP/5060-b5b13340", "SIP/23003") in new stack Audio is at 89.35.79.5 port 16240 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 89.35.79.69:39246: INVITE sip:23003@89.35.79.69:39246;rinstance=a9cce79c2a37b77b SIP/2.0 Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK6a3c2910;rport From: "TEST" ;tag=as46e6ae2d To: Contact: Call-ID: 49eac613368d6f0e1fc464a63cd65ee4@89.35.79.5 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 05 Mar 2007 02:27:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 251 v=0 o=root 6308 6308 IN IP4 10.0.0.1 s=session c=IN IP4 10.0.0.1 t=0 0 m=audio 10000 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 23003 voip*CLI> <--- SIP read from 89.35.79.69:39246 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK6a3c2910;rport=5060 Contact: To: ;tag=b9473829 From: "TEST";tag=as46e6ae2d Call-ID: 49eac613368d6f0e1fc464a63cd65ee4@89.35.79.5 CSeq: 102 INVITE User-Agent: X-Lite release 1006e stamp 34025 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- -- SIP/23003-08277f78 is ringing <--- Transmitting (no NAT) to 10.0.0.1:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-970-24fb54-3488;received=10.0.0.1 From: "TEST";tag=a000001-13c4-970-24fb54-382b To: ;tag=as788a2f98 Call-ID: 9794353c-a000001-13c4-970-24fb54-3ca5@10.0.0.1 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> voip*CLI> <--- SIP read from 10.0.0.1:5060 ---> SIP/2.0 415 Unsupported Media Type From: "Test";tag=as67d70ee4 To: ;tag=a000001-13c4-96f-24f9c8-a0f Call-ID: 47b5de4a4d5113581f9bc6c81d2becb7@10.0.0.2 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.0.0.2:5060;received=10.0.0.2;branch=z9hG4bK05b1673f;rport Content-Length:0 <-------------> --- (7 headers 0 lines) --- -- Got SIP response 415 "Unsupported Media Type" back from 10.0.0.1 Transmitting (no NAT) to 10.0.0.1:5060: ACK sip:23003@10.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK05b1673f;rport From: "Test" ;tag=as67d70ee4 To: ;tag=a000001-13c4-96f-24f9c8-a0f Contact: Call-ID: 47b5de4a4d5113581f9bc6c81d2becb7@10.0.0.2 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- == Everyone is busy/congested at this time (1:0/0/1) -- Executing [s@macro-trunkdial:2] Goto("SIP/23001-b5b0f3d8", "s-CHANUNAVAIL|1") in new stack -- Goto (macro-trunkdial,s-CHANUNAVAIL,1) -- Executing [s-CHANUNAVAIL@macro-trunkdial:1] NoOp("SIP/23001-b5b0f3d8", "") in new stack voip*CLI> <--- SIP read from 10.0.0.1:5060 ---> CANCEL sip:23003@10.0.0.2;transport=UDP;TCID-27 SIP/2.0 From: ;tag=a000001-13c4-970-24fb54-382b To: Call-ID: 9794353c-a000001-13c4-970-24fb54-3ca5@10.0.0.1 CSeq: 1 CANCEL Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-970-24fb54-3488 Max-Forwards: 70 Content-Length:0 <-------------> --- (8 headers 0 lines) --- Sending to 10.0.0.1 : 5060 (no NAT) <--- Reliably Transmitting (no NAT) to 10.0.0.1:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-970-24fb54-3488;received=10.0.0.1 From: "TEST";tag=a000001-13c4-970-24fb54-382b To: ;tag=as788a2f98 Call-ID: 9794353c-a000001-13c4-970-24fb54-3ca5@10.0.0.1 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> <--- Transmitting (no NAT) to 10.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-970-24fb54-3488;received=10.0.0.1 From: ;tag=a000001-13c4-970-24fb54-382b To: ;tag=as788a2f98 Call-ID: 9794353c-a000001-13c4-970-24fb54-3ca5@10.0.0.1 CSeq: 1 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Really destroying SIP dialog '47b5de4a4d5113581f9bc6c81d2becb7@10.0.0.2' Method: INVITE Scheduling destruction of SIP dialog '49eac613368d6f0e1fc464a63cd65ee4@89.35.79.5' in 32000 ms (Method: INVITE) Reliably Transmitting (no NAT) to 89.35.79.69:39246: CANCEL sip:23003@89.35.79.69:39246;rinstance=a9cce79c2a37b77b SIP/2.0 Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK6a3c2910;rport From: "TEST" ;tag=as46e6ae2d To: Call-ID: 49eac613368d6f0e1fc464a63cd65ee4@89.35.79.5 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '49eac613368d6f0e1fc464a63cd65ee4@89.35.79.5' in 32000 ms (Method: INVITE) == Spawn extension (DID_trunk_1, 23003, 1) exited non-zero on 'SIP/5060-b5b13340' voip*CLI> <--- SIP read from 10.0.0.1:5060 ---> ACK sip:23003@10.0.0.2;transport=UDP;TCID-27 SIP/2.0 From: ;tag=a000001-13c4-970-24fb54-382b To: ;tag=as788a2f98 Call-ID: 9794353c-a000001-13c4-970-24fb54-3ca5@10.0.0.1 CSeq: 1 ACK Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-970-24fb54-3488 Max-Forwards: 70 Contact: Content-Length:0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '9794353c-a000001-13c4-970-24fb54-3ca5@10.0.0.1' Method: ACK voip*CLI> <--- SIP read from 89.35.79.69:39246 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK6a3c2910;rport=5060 Contact: To: ;tag=b9473829 From: "TEST";tag=as46e6ae2d Call-ID: 49eac613368d6f0e1fc464a63cd65ee4@89.35.79.5 CSeq: 102 CANCEL User-Agent: X-Lite release 1006e stamp 34025 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- voip*CLI> <--- SIP read from 89.35.79.69:39246 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK6a3c2910;rport=5060 To: ;tag=b9473829 From: "TEST";tag=as46e6ae2d Call-ID: 49eac613368d6f0e1fc464a63cd65ee4@89.35.79.5 CSeq: 102 INVITE User-Agent: X-Lite release 1006e stamp 34025 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (no NAT) to 89.35.79.69:39246: ACK sip:23003@89.35.79.69:39246;rinstance=a9cce79c2a37b77b SIP/2.0 Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK6a3c2910;rport From: "TEST" ;tag=as46e6ae2d To: ;tag=b9473829 Contact: Call-ID: 49eac613368d6f0e1fc464a63cd65ee4@89.35.79.5 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- voip*CLI> <--- SIP read from 89.35.79.247:5060 ---> CANCEL sip:23003@voip.telefonet.ro SIP/2.0 Via: SIP/2.0/UDP 89.35.79.247:5060;branch=z9hG4bKfS4gesXYv Max-Forwards: 70 User-Agent: Wu Chuan From: "23001" ;tag=YG6UP1oYxeS4ilVp To: "23003" Call-ID: SPbbitaAMsMutV8Q@89.35.79.247 Contact: Proxy-Authorization: Digest username="23001", realm="asterisk", nonce="25942128", uri="sip:23003@voip.telefonet.ro", response="b12da7f6a02fc62a45fe9453d928853d" CSeq: 2 CANCEL Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 89.35.79.247 : 5060 (no NAT) <--- Reliably Transmitting (no NAT) to 89.35.79.247:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 89.35.79.247:5060;branch=z9hG4bKfS4gesXYv;received=89.35.79.247 From: "23001" ;tag=YG6UP1oYxeS4ilVp To: "23003" ;tag=as5ba87637 Call-ID: SPbbitaAMsMutV8Q@89.35.79.247 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> <--- Transmitting (no NAT) to 89.35.79.247:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 89.35.79.247:5060;branch=z9hG4bKfS4gesXYv;received=89.35.79.247 From: "23001" ;tag=YG6UP1oYxeS4ilVp To: "23003" ;tag=as5ba87637 Call-ID: SPbbitaAMsMutV8Q@89.35.79.247 CSeq: 2 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> voip*CLI> <--- SIP read from 89.35.79.247:5060 ---> ACK sip:23003@voip.telefonet.ro SIP/2.0 Via: SIP/2.0/UDP 89.35.79.247:5060;branch=z9hG4bKfS4gesXYv Max-Forwards: 70 User-Agent: Wu Chuan From: "23001" ;tag=YG6UP1oYxeS4ilVp To: "23003" Call-ID: SPbbitaAMsMutV8Q@89.35.79.247 Contact: Proxy-Authorization: Digest username="23001", realm="asterisk", nonce="25942128", uri="sip:23003@voip.telefonet.ro", response="b12da7f6a02fc62a45fe9453d928853d" CSeq: 2 ACK Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog 'SPbbitaAMsMutV8Q@89.35.79.247' Method: ACK voip*CLI>