=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2007.11.28 12:58:34 =~=~=~=~=~=~=~=~=~=~=~= asterisk2*CLI> asterisk2*CLI> asterisk2*CLI> asterisk2*CLI> asterisk2*CLI> sip debug peer andrew_ata1_line1 asterisk2*CLI> SIP Debugging Enabled for IP: 24.155.147.204:5060 asterisk2*CLI> <--- SIP read from 24.155.147.204:5060 ---> INVITE sip:1691@72.48.145.6 SIP/2.0 Via: SIP/2.0/UDP 24.155.147.204:5060;branch=z9hG4bK-21fc83c8 From: andrew_ata1_line1 ;tag=dea7d0b4ed9b78e8o0 To: Call-ID: 4f1095a4-c987f8b8@24.155.147.204 CSeq: 101 INVITE Max-Forwards: 70 Contact: andrew_ata1_line1 Expires: 240 User-Agent: Sipura/SPA2002-3.1.5 Content-Length: 428 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 7667597 7667597 IN IP4 24.155.147.204 s=- c=IN IP4 24.155.147.204 t=0 0 m=audio 16442 RTP/AVP 18 0 2 4 8 96 97 98 100 101 a=rtpmap:18 G729a/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/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 asterisk2*CLI> <-------------> --- (14 headers 19 lines) --- Sending to 24.155.147.204 : 5060 (no NAT) Using INVITE request as basis request - 4f1095a4-c987f8b8@24.155.147.204 <--- Reliably Transmitting (no NAT) to 24.155.147.204:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 24.155.147.204:5060;branch=z9hG4bK-21fc83c8;received=24.155.147.204 From: andrew_ata1_line1 ;tag=dea7d0b4ed9b78e8o0 To: ;tag=as11d4820a Call-ID: 4f1095a4-c987f8b8@24.155.147.204 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4c9a7ccb" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '4f1095a4-c987f8b8@24.155.147.204' in 32000 ms (Method: INVITE) Found user 'andrew_ata1_line1' asterisk2*CLI> <--- SIP read from 24.155.147.204:5060 ---> ACK sip:1691@72.48.145.6 SIP/2.0 Via: SIP/2.0/UDP 24.155.147.204:5060;branch=z9hG4bK-21fc83c8 From: andrew_ata1_line1 ;tag=dea7d0b4ed9b78e8o0 To: ;tag=as11d4820a Call-ID: 4f1095a4-c987f8b8@24.155.147.204 CSeq: 101 ACK Max-Forwards: 70 Contact: andrew_ata1_line1 User-Agent: Sipura/SPA2002-3.1.5 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk2*CLI> <--- SIP read from 24.155.147.204:5060 ---> INVITE sip:1691@72.48.145.6 SIP/2.0 Via: SIP/2.0/UDP 24.155.147.204:5060;branch=z9hG4bK-9b3b7660 From: andrew_ata1_line1 ;tag=dea7d0b4ed9b78e8o0 To: Call-ID: 4f1095a4-c987f8b8@24.155.147.204 CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="andrew_ata1_line1",realm="asterisk",nonce="4c9a7ccb",uri="sip:1691@72.48.145.6",algorithm=MD5,response="bd6e732d39a570b236e447c591f15974" Contact: andrew_ata1_line1 Expires: 240 User-Agent: Sipura/SPA2002-3.1.5 Content-Length: 428 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 7667597 7667597 IN IP4 24.155.147.204 s=- c=IN IP4 24.155.147.204 t=0 0 m=audio 16442 RTP/AVP 18 0 2 4 8 96 97 98 100 101 a=rtpmap:18 G729a/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a asterisk2*CLI> =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 <-------------> --- (15 headers 19 lines) --- Sending to 24.155.147.204 : 5060 (no NAT) Using INVITE request as basis request - 4f1095a4-c987f8b8@24.155.147.204 Found user 'andrew_ata1_line1' Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 2 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 96 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 100 Found RTP audio format 101 Peer audio RTP is at port 24.155.147.204:16442 Found description format G729a for ID 18 Found description format PCMU for ID 0 Found description format G726-32 for ID 2 Found description format G723 for ID 4 Found description format PCMA for ID 8 Found description format G726-40 for ID 96 Found description format G726-24 for ID 97 Found description format G726-16 for ID 98 Found description format NSE for ID 100 Found description format telephone-event for ID 101 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd0d (g723|ulaw|alaw|g726|g729|ilbc)/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 24.155.147.204:16442 Looking for 1691 in outbound (domain 72.48.145.6) list_route: hop: <--- Transmitting (no NAT) to 24.155.147.204:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 24.155.147.204:5060;branch=z9hG4bK-9b3b7660;received=24.155.147.204 From: andrew_ata1_line1 ;tag=dea7d0b4ed9b78e8o0 To: Call-ID: 4f1095a4-c987f8b8@24.155.147.204 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> asterisk2*CLI> <--- Transmitting (no NAT) to 24.155.147.204:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 24.155.147.204:5060;branch=z9hG4bK-9b3b7660;received=24.155.147.204 From: andrew_ata1_line1 ;tag=dea7d0b4ed9b78e8o0 To: ;tag=as40a82c1e Call-ID: 4f1095a4-c987f8b8@24.155.147.204 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> asterisk2*CLI> <--- SIP read from 24.155.147.204:5060 ---> INFO sip:1691@72.48.145.6 SIP/2.0 Via: SIP/2.0/UDP 24.155.147.204:5060;branch=z9hG4bK-b25e30fb From: andrew_ata1_line1 ;tag=dea7d0b4ed9b78e8o0 To: Call-ID: 4f1095a4-c987f8b8@24.155.147.204 CSeq: 103 INFO Max-Forwards: 70 Proxy-Authorization: Digest username="andrew_ata1_line1",realm="asterisk",nonce="4c9a7ccb",uri="sip:1691@72.48.145.6",algorithm=MD5,response="8d81a8bbfaaf2a93e0c821c6871f37d0" User-Agent: Sipura/SPA2002-3.1.5 Content-Length: 24 Content-Type: application/dtmf-relay Signal=5 Duration=100 <-------------> --- (11 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: 5 <--- Transmitting (no NAT) to 24.155.147.204:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 24.155.147.204:5060;branch=z9hG4bK-b25e30fb;received=24.155.147.204 From: andrew_ata1_line1 ;tag=dea7d0b4ed9b78e8o0 To: ;tag=as40a82c1e Call-ID: 4f1095a4-c987f8b8@24.155.147.204 CSeq: 103 INFO User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> asterisk2*CLI> <--- SIP read from 24.155.147.204:5060 ---> CANCEL sip:1691@72.48.145.6 SIP/2.0 Via: SIP/2.0/UDP 24.155.147.204:5060;branch=z9hG4bK-9b3b7660 From: andrew_ata1_line1 ;tag=dea7d0b4ed9b78e8o0 To: Call-ID: 4f1095a4-c987f8b8@24.155.147.204 CSeq: 102 CANCEL Max-Forwards: 70 Proxy-Authorization: Digest username="andrew_ata1_line1",realm="asterisk",nonce="4c9a7ccb",uri="sip:1691@72.48.145.6",algorithm=MD5,response="63941307978abf1d5680b38c440ee020" User-Agent: Sipura/SPA2002-3.1.5 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- Transmitting (no NAT) to 24.155.147.204:5060 ---> SIP/2.0 503 Server error Via: SIP/2.0/UDP 24.155.147.204:5060;branch=z9hG4bK-9b3b7660;received=24.155.147.204 From: andrew_ata1_line1 ;tag=dea7d0b4ed9b78e8o0 To: ;tag=as40a82c1e Call-ID: 4f1095a4-c987f8b8@24.155.147.204 CSeq: 102 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> asterisk2*CLI> Reliably Transmitting (no NAT) to 24.155.147.204:5060: OPTIONS sip:andrew_ata1_line1@24.155.147.204:5060 SIP/2.0 Via: SIP/2.0/UDP 72.48.145.6:5060;branch=z9hG4bK4e528ea2;rport From: "asterisk" ;tag=as1a9eed41 To: Contact: Call-ID: 07d51230546f8efb5259e7a62a49b636@72.48.145.6 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 28 Nov 2007 17:59:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- asterisk2*CLI> <--- SIP read from 24.155.147.204:5060 ---> SIP/2.0 200 OK To: ;tag=e2df5428d6ddad06i0 From: "asterisk" ;tag=as1a9eed41 Call-ID: 07d51230546f8efb5259e7a62a49b636@72.48.145.6 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 72.48.145.6:5060;branch=z9hG4bK4e528ea2 Server: Sipura/SPA2002-3.1.5 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '07d51230546f8efb5259e7a62a49b636@72.48.145.6' Method: OPTIONS asterisk2*CLI> [Nov 28 12:00:18] NOTICE[5597]: cdr.c:434 ast_cdr_free: CDR on channel 'SIP/andrew_ata2-08c8e578' not posted <--- Transmitting (no NAT) to 24.155.147.204:5060 ---> SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 24.155.147.204:5060;branch=z9hG4bK-9b3b7660;received=24.155.147.204 From: andrew_ata1_line1 ;tag=dea7d0b4ed9b78e8o0 To: ;tag=as40a82c1e Call-ID: 4f1095a4-c987f8b8@24.155.147.204 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> asterisk2*CLI> Really destroying SIP dialog '4f1095a4-c987f8b8@24.155.147.204' Method: CANCEL asterisk2*CLI> Reliably Transmitting (no NAT) to 24.155.147.204:5060: OPTIONS sip:andrew_ata1_line1@24.155.147.204:5060 SIP/2.0 Via: SIP/2.0/UDP 72.48.145.6:5060;branch=z9hG4bK55259d83;rport From: "asterisk" ;tag=as4cb106e3 To: Contact: Call-ID: 4ff153393390637b3da7b73a0527bcff@72.48.145.6 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 28 Nov 2007 18:00:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- asterisk2*CLI> <--- SIP read from 24.155.147.204:5060 ---> SIP/2.0 200 OK To: ;tag=e2df5428d6ddad06i0 From: "asterisk" ;tag=as4cb106e3 Call-ID: 4ff153393390637b3da7b73a0527bcff@72.48.145.6 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 72.48.145.6:5060;branch=z9hG4bK55259d83 Server: Sipura/SPA2002-3.1.5 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '4ff153393390637b3da7b73a0527bcff@72.48.145.6' Method: OPTIONS asterisk2*CLI>