Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 89.35.79.69:56404 Looking for 0748881658 in numberplan-custom-1 (domain voip.telefonet.ro) list_route: hop: voip*CLI> <--- Transmitting (no NAT) to 89.35.79.69:61286 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 89.35.79.69:61286;branch=z9hG4bK-d87543-41418932c6173f26-1--d87543-;received=89.35.79.69;rport=61286 From: "Znuff";tag=856ae020 To: "0748881658" Call-ID: ZjhkNjdlYTFiN2NkNWNmNDJhZTgxMzZiYWJkZDEwMTU. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [0748881658@numberplan-custom-1:1] Dial("SIP/23003-b5a1c108", "SIP/0748881658@10.0.0.1") in new stack Audio is at 10.0.0.2 port 15592 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:0748881658@10.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK3542a8a3;rport From: "Znuff" ;tag=as69b93c0a To: Contact: Call-ID: 565304bb089b4fff4719b03c311ec869@10.0.0.2 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 05 Mar 2007 04:21:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 236 v=0 o=root 8856 8856 IN IP4 89.35.79.69 s=session c=IN IP4 89.35.79.69 t=0 0 m=audio 56404 RTP/AVP 8 101 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 0748881658@10.0.0.1 voip*CLI> <--- SIP read from 10.0.0.1:5060 ---> SIP/2.0 100 Trying From: "Znuff";tag=as69b93c0a To: ;tag=a000001-13c4-2413-8d07c6-36c1 Call-ID: 565304bb089b4fff4719b03c311ec869@10.0.0.2 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.0.0.2:5060;received=10.0.0.2;branch=z9hG4bK3542a8a3;rport Contact: Content-Length:0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog 'XfLu8uaWvRwQUWiM@89.35.79.247' Method: REGISTER voip*CLI> <--- SIP read from 10.0.0.1:5060 ---> SIP/2.0 415 Unsupported Media Type From: "Znuff";tag=as69b93c0a To: ;tag=a000001-13c4-2413-8d07c6-36c1 Call-ID: 565304bb089b4fff4719b03c311ec869@10.0.0.2 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.0.0.2:5060;received=10.0.0.2;branch=z9hG4bK3542a8a3;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:0748881658@10.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK3542a8a3;rport From: "Znuff" ;tag=as69b93c0a To: ;tag=a000001-13c4-2413-8d07c6-36c1 Contact: Call-ID: 565304bb089b4fff4719b03c311ec869@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 [0748881658@numberplan-custom-1:2] Congestion("SIP/23003-b5a1c108", "") in new stack voip*CLI> <--- Transmitting (no NAT) to 89.35.79.69:61286 ---> SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 89.35.79.69:61286;branch=z9hG4bK-d87543-41418932c6173f26-1--d87543-;received=89.35.79.69;rport=61286 From: "Znuff";tag=856ae020 To: "0748881658";tag=as6f87b067 Call-ID: ZjhkNjdlYTFiN2NkNWNmNDJhZTgxMzZiYWJkZDEwMTU. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 X-Asterisk-HangupCause: Interworking, unspecified X-Asterisk-HangupCauseCode: 127 <------------> == Spawn extension (numberplan-custom-1, 0748881658, 2) exited non-zero on 'SIP/23003-b5a1c108' voip*CLI> <--- SIP read from 89.35.79.69:61286 ---> ACK sip:0748881658@voip.telefonet.ro SIP/2.0 Via: SIP/2.0/UDP 89.35.79.69:61286;branch=z9hG4bK-d87543-41418932c6173f26-1--d87543-;rport To: "0748881658";tag=as6f87b067 From: "Znuff";tag=856ae020 Call-ID: ZjhkNjdlYTFiN2NkNWNmNDJhZTgxMzZiYWJkZDEwMTU. CSeq: 2 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '565304bb089b4fff4719b03c311ec869@10.0.0.2' Method: INVITE Really destroying SIP dialog 'ZjhkNjdlYTFiN2NkNWNmNDJhZTgxMzZiYWJkZDEwMTU.' Method: ACK voip*CLI> <--- SIP read from 89.35.79.69:61286 ---> <-------------> --- (0 headers 1 lines) --- Really destroying SIP dialog 'owiz95Hz9XJxvP8b@89.35.79.94' Method: REGISTER voip*CLI> <--- SIP read from 89.35.79.247:5060 ---> REGISTER sip:voip.telefonet.ro SIP/2.0 Via: SIP/2.0/UDP 89.35.79.247:5060;branch=z9hG4bKIzCib2opx Max-Forwards: 70 User-Agent: Wu Chuan From: "23001" ;tag=hOC1H8UNFOCvPuLu To: "23001" Call-ID: XfLu8uaWvRwQUWiM@89.35.79.247 CSeq: 9806 REGISTER Contact: Expires: 60 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 89.35.79.247 : 5060 (no NAT) voip*CLI> <--- 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=z9hG4bKIzCib2opx;received=89.35.79.247 From: "23001" ;tag=hOC1H8UNFOCvPuLu To: "23001" Call-ID: XfLu8uaWvRwQUWiM@89.35.79.247 CSeq: 9806 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 89.35.79.247:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 89.35.79.247:5060;branch=z9hG4bKIzCib2opx;received=89.35.79.247 From: "23001" ;tag=hOC1H8UNFOCvPuLu To: "23001" ;tag=as70d42adb Call-ID: XfLu8uaWvRwQUWiM@89.35.79.247 CSeq: 9806 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="voip.telefonet.ro", nonce="31195ebf" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'XfLu8uaWvRwQUWiM@89.35.79.247' in 32000 ms (Method: REGISTER) voip*CLI> <--- SIP read from 89.35.79.247:5060 ---> REGISTER sip:voip.telefonet.ro SIP/2.0 Via: SIP/2.0/UDP 89.35.79.247:5060;branch=z9hG4bKdNux7SGxR Max-Forwards: 70 User-Agent: Wu Chuan From: "23001" ;tag=YhLcODA3FkkSvvdi To: "23001" Call-ID: XfLu8uaWvRwQUWiM@89.35.79.247 CSeq: 9807 REGISTER Contact: Expires: 60 Authorization: Digest username="23001", realm="voip.telefonet.ro", nonce="31195ebf", uri="sip:voip.telefonet.ro", response="b7406079f2061834b4245da55f0a0a26" Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 89.35.79.247 : 5060 (no NAT) voip*CLI> <--- 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=z9hG4bKdNux7SGxR;received=89.35.79.247 From: "23001" ;tag=YhLcODA3FkkSvvdi To: "23001" Call-ID: XfLu8uaWvRwQUWiM@89.35.79.247 CSeq: 9807 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 89.35.79.247:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 89.35.79.247:5060;branch=z9hG4bKdNux7SGxR;received=89.35.79.247 From: "23001" ;tag=YhLcODA3FkkSvvdi To: "23001" ;tag=as70d42adb Call-ID: XfLu8uaWvRwQUWiM@89.35.79.247 CSeq: 9807 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 04:21:47 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'XfLu8uaWvRwQUWiM@89.35.79.247' in 32000 ms (Method: REGISTER) voip*CLI>