shaun*CLI> sip set debug peer 3001 SIP Debugging Enabled for IP: 217.37.44.41:56541 shaun*CLI> <--- SIP read from 217.37.44.41:56541 ---> <-------------> --- (0 headers 1 lines) --- shaun*CLI> <--- SIP read from 217.37.44.41:56541 ---> INVITE sip:901296991234@garysoft.co.uk SIP/2.0 Via: SIP/2.0/UDP 10.10.1.129:34964;branch=z9hG4bK-d87543-03220078dd0a0235-1--d87543-;rport Max-Forwards: 70 Contact: To: "901296991234" From: "Gary Hawkins";tag=a752e74a Call-ID: MjUzODU2NTkwMDAyNGIzNmY0Nzc3MTFhZDdlZGY4YjA. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1006e stamp 34025 Content-Length: 230 v=0 o=- 2 2 IN IP4 10.10.1.129 s=CounterPath X-Lite 3.0 c=IN IP4 10.10.1.129 t=0 0 m=audio 23418 RTP/AVP 3 101 a=alt:1 1 : w5YTrm84 XzK+0maV 10.10.1.129 23418 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> --- (12 headers 10 lines) --- Sending to 217.37.44.41 : 56541 (NAT) Using INVITE request as basis request - MjUzODU2NTkwMDAyNGIzNmY0Nzc3MTFhZDdlZGY4YjA. <--- Reliably Transmitting (NAT) to 217.37.44.41:56541 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.10.1.129:34964;branch=z9hG4bK-d87543-03220078dd0a0235-1--d87543-;received=217.37.44.41;rport=56541 From: "Gary Hawkins";tag=a752e74a To: "901296991234";tag=as11c6384f Call-ID: MjUzODU2NTkwMDAyNGIzNmY0Nzc3MTFhZDdlZGY4YjA. CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="garysoft.co.uk", nonce="3c38f563" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'MjUzODU2NTkwMDAyNGIzNmY0Nzc3MTFhZDdlZGY4YjA.' in 32000 ms (Method: INVITE) Found user '3001' shaun*CLI> <--- SIP read from 217.37.44.41:56541 ---> ACK sip:901296991234@garysoft.co.uk SIP/2.0 Via: SIP/2.0/UDP 10.10.1.129:34964;branch=z9hG4bK-d87543-03220078dd0a0235-1--d87543-;rport To: "901296991234";tag=as11c6384f From: "Gary Hawkins";tag=a752e74a Call-ID: MjUzODU2NTkwMDAyNGIzNmY0Nzc3MTFhZDdlZGY4YjA. CSeq: 1 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- shaun*CLI> <--- SIP read from 217.37.44.41:56541 ---> INVITE sip:901296991234@garysoft.co.uk SIP/2.0 Via: SIP/2.0/UDP 10.10.1.129:34964;branch=z9hG4bK-d87543-e121c25b1d2c7346-1--d87543-;rport Max-Forwards: 70 Contact: To: "901296991234" From: "Gary Hawkins";tag=a752e74a Call-ID: MjUzODU2NTkwMDAyNGIzNmY0Nzc3MTFhZDdlZGY4YjA. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Proxy-Authorization: Digest username="3001",realm="garysoft.co.uk",nonce="3c38f563",uri="sip:901296991234@garysoft.co.uk",response="cefb026e242ffbc16c6367d81c635ff2",algorithm=MD5 User-Agent: X-Lite release 1006e stamp 34025 Content-Length: 230 v=0 o=- 2 2 IN IP4 10.10.1.129 s=CounterPath X-Lite 3.0 c=IN IP4 10.10.1.129 t=0 0 m=audio 23418 RTP/AVP 3 101 a=alt:1 1 : w5YTrm84 XzK+0maV 10.10.1.129 23418 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv -------------> --- (13 headers 10 lines) --- Sending to 217.37.44.41 : 56541 (NAT) Using INVITE request as basis request - MjUzODU2NTkwMDAyNGIzNmY0Nzc3MTFhZDdlZGY4YjA. Found user '3001' Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 10.10.1.129:23418 Found description format telephone-event for ID 101 Capabilities: us - 0xc02 (gsm|g726|ilbc), peer - audio=0x2 (gsm)/video=0x0 (nothing), combined - 0x2 (gsm) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.10.1.129:23418 Looking for 901296991234 in sip-outgoing (domain garysoft.co.uk) list_route: hop: <--- Transmitting (NAT) to 217.37.44.41:56541 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.1.129:34964;branch=z9hG4bK-d87543-e121c25b1d2c7346-1--d87543-;received=217.37.44.41;rport=56541 From: "Gary Hawkins";tag=a752e74a To: "901296991234" Call-ID: MjUzODU2NTkwMDAyNGIzNmY0Nzc3MTFhZDdlZGY4YjA. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> shaun*CLI> <--- Transmitting (NAT) to 217.37.44.41:56541 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.1.129:34964;branch=z9hG4bK-d87543-e121c25b1d2c7346-1--d87543-;received=217.37.44.41;rport=56541 From: "Gary Hawkins";tag=a752e74a To: "901296991234" Call-ID: MjUzODU2NTkwMDAyNGIzNmY0Nzc3MTFhZDdlZGY4YjA. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Audio is at 81.2.105.35 port 27330 Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (NAT) to 217.37.44.41:56541 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.10.1.129:34964;branch=z9hG4bK-d87543-e121c25b1d2c7346-1--d87543-;received=217.37.44.41;rport=56541 From: "Gary Hawkins";tag=a752e74a To: "901296991234";tag=as5d80ab09 Call-ID: MjUzODU2NTkwMDAyNGIzNmY0Nzc3MTFhZDdlZGY4YjA. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 235 v=0 o=root 7616 7616 IN IP4 81.2.105.35 s=session c=IN IP4 81.2.105.35 t=0 0 m=audio 27330 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> Scheduling destruction of SIP dialog 'MjUzODU2NTkwMDAyNGIzNmY0Nzc3MTFhZDdlZGY4YjA.' in 32000 ms (Method: INVITE) <--- Reliably Transmitting (NAT) to 217.37.44.41:56541 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.1.129:34964;branch=z9hG4bK-d87543-e121c25b1d2c7346-1--d87543-;received=217.37.44.41;rport=56541 From: "Gary Hawkins";tag=a752e74a To: "901296991234";tag=as5d80ab09 Call-ID: MjUzODU2NTkwMDAyNGIzNmY0Nzc3MTFhZDdlZGY4YjA. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> shaun*CLI> <--- SIP read from 217.37.44.41:56541 ---> ACK sip:901296991234@garysoft.co.uk SIP/2.0 Via: SIP/2.0/UDP 10.10.1.129:34964;branch=z9hG4bK-d87543-e121c25b1d2c7346-1--d87543-;rport To: "901296991234";tag=as5d80ab09 From: "Gary Hawkins";tag=a752e74a Call-ID: MjUzODU2NTkwMDAyNGIzNmY0Nzc3MTFhZDdlZGY4YjA. CSeq: 2 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Reliably Transmitting (NAT) to 217.37.44.41:56541: OPTIONS sip:3001@217.37.44.41:56541;rinstance=1bf00f2ebc5aa6f8 SIP/2.0 Via: SIP/2.0/UDP 81.2.105.35:5060;branch=z9hG4bK16c06d1c;rport From: "asterisk" ;tag=as3768180c To: Contact: Call-ID: 3fe58f254f164bfb306b9da50cf2086b@garysoft.co.uk CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 23 Mar 2007 13:09:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- shaun*CLI> <--- SIP read from 217.37.44.41:56541 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 81.2.105.35:5060;branch=z9hG4bK16c06d1c;rport=5060 Contact: To: ;tag=dd4e4d1c From: "asterisk";tag=as3768180c Call-ID: 3fe58f254f164bfb306b9da50cf2086b@garysoft.co.uk CSeq: 102 OPTIONS Accept: application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1006e stamp 34025 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '3fe58f254f164bfb306b9da50cf2086b@garysoft.co.uk' Method: OPTIONS shaun*CLI>