*CLI> Reliably Transmitting (NAT) to 65.175.129.133:5060: OPTIONS sip:proxy2.bandtel.com SIP/2.0 Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK22cfd3c4;rport From: "asterisk" ;tag=as7d7f766d To: Contact: Call-ID: 111b5688643541676b70b1552f812a5b@10.1.1.68 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 04 Dec 2007 09:56:14 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- <--- SIP read from 65.175.129.133:5060 ---> SIP/2.0 501 Not Implemented Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK22cfd3c4;rport=5060 From: asterisk ;tag=as7d7f766d To: Call-ID: 111b5688643541676b70b1552f812a5b@10.1.1.68 CSeq: 102 OPTIONS Server: Sippy <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '111b5688643541676b70b1552f812a5b@10.1.1.68' Method: OPTIONS -- Attempting call on Local/outbound@dialout for outbound-handler@dialout:1 (Retry 1) -- Executing [outbound@dialout:1] Answer("Local/outbound@dialout-78ba,2", "") in new stack -- Executing [outbound@dialout:2] Wait("Local/outbound@dialout-78ba,2", "50") in new stack -- Executing [outbound-handler@dialout:1] Dial("Local/outbound@dialout-78ba,1", "SIP/011919960466622@proxy2.bandtel.com|70|gM(outbound-connect^agi://10.1.1.68/ivr/speak^---+%0Aname%3A+sanchu%0Aid%3A+1%0A^)") in new stack Audio is at 10.1.1.68 port 10988 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 65.175.129.133:5060: INVITE sip:011919960466622@proxy2.bandtel.com SIP/2.0 Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK5c12b644;rport From: "2068200001" ;tag=as6de88472 To: Contact: Call-ID: 3c0b902d3343200937839d954e419945@proxy2.bandtel.com CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "2068200001" ;privacy=off;screen=no Date: Tue, 04 Dec 2007 09:56:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 234 v=0 o=root 30571 30571 IN IP4 10.1.1.68 s=session c=IN IP4 10.1.1.68 t=0 0 m=audio 10988 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 011919960466622@proxy2.bandtel.com <--- SIP read from 65.175.129.133:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK5c12b644;rport=5060 From: "2068200001" ;tag=as6de88472 To: Call-ID: 3c0b902d3343200937839d954e419945@proxy2.bandtel.com CSeq: 102 INVITE Server: Sip EXpress router (0.9.6 (i386/freebsd)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 65.175.129.133:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK5c12b644;rport=5060 Record-Route: From: 2068200001 ;tag=as6de88472 To: Call-ID: 3c0b902d3343200937839d954e419945@proxy2.bandtel.com CSeq: 102 INVITE Server: Sippy WWW-Authenticate: Digest realm="65.175.129.133",nonce="ec13ba9157d5c517c1efb05fc30fe17247552714" <-------------> --- (9 headers 0 lines) --- Transmitting (NAT) to 65.175.129.133:5060: ACK sip:011919960466622@proxy2.bandtel.com SIP/2.0 Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK5c12b644;rport From: "2068200001" ;tag=as6de88472 To: Contact: Call-ID: 3c0b902d3343200937839d954e419945@proxy2.bandtel.com CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "2068200001" ;privacy=off;screen=no Content-Length: 0 --- Audio is at 10.1.1.68 port 10988 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 65.175.129.133:5060: INVITE sip:011919960466622@proxy2.bandtel.com SIP/2.0 Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK658d6d72;rport From: "2068200001" ;tag=as6de88472 To: Contact: Call-ID: 3c0b902d3343200937839d954e419945@proxy2.bandtel.com CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "2068200001" ;privacy=off;screen=no Authorization: Digest username="2068200001", realm="65.175.129.133", algorithm=MD5, uri="sip:011919960466622@proxy2.bandtel.com", nonce="ec13ba9157d5c517c1efb05fc30fe17247552714", response="2511411b6a656bedf21d7549795c2e1a", opaque="" Date: Tue, 04 Dec 2007 09:56:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 234 v=0 o=root 30571 30572 IN IP4 10.1.1.68 s=session c=IN IP4 10.1.1.68 t=0 0 m=audio 10988 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- <--- SIP read from 65.175.129.133:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK658d6d72;rport=5060 From: "2068200001" ;tag=as6de88472 To: Call-ID: 3c0b902d3343200937839d954e419945@proxy2.bandtel.com CSeq: 103 INVITE Server: Sip EXpress router (0.9.6 (i386/freebsd)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 65.175.129.133:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK658d6d72;rport=5060 Record-Route: From: 2068200001 ;tag=as6de88472 To: ;tag=2a3ad5978d87b904137d93d2eef8307b Call-ID: 3c0b902d3343200937839d954e419945@proxy2.bandtel.com CSeq: 103 INVITE Server: Sippy Content-Length: 169 Content-Type: application/sdp v=0 o=prxlax01 0 0 IN IP4 216.168.169.108 s=sip call t=0 0 m=audio 19206 RTP/AVP 0 101 c=IN IP4 216.168.169.87 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (10 headers 8 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 216.168.169.87:19206 Found description format telephone-event for ID 101 Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 216.168.169.87:19206 -- SIP/proxy2.bandtel.com-08ffb4f0 is making progress passing it to Local/outbound@dialout-78ba,1 Reliably Transmitting (NAT) to 65.175.129.133:5060: OPTIONS sip:proxy2.bandtel.com SIP/2.0 Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK0f2bd66a;rport From: "asterisk" ;tag=as2183c12e To: Contact: Call-ID: 038483a862b18f72336099c06b716996@10.1.1.68 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 04 Dec 2007 09:57:14 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- <--- SIP read from 65.175.129.133:5060 ---> SIP/2.0 501 Not Implemented Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK0f2bd66a;rport=5060 From: asterisk ;tag=as2183c12e To: Call-ID: 038483a862b18f72336099c06b716996@10.1.1.68 CSeq: 102 OPTIONS Server: Sippy <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '038483a862b18f72336099c06b716996@10.1.1.68' Method: OPTIONS -- Executing [outbound@dialout:3] NoOp("Local/outbound@dialout-78ba,2", "status=") in new stack -- Executing [outbound@dialout:4] AGI("Local/outbound@dialout-78ba,2", "agi://10.1.1.68/ivr/unanswered") in new stack -- AGI Script agi://10.1.1.68/ivr/unanswered completed, returning 0 -- Executing [outbound@dialout:5] Hangup("Local/outbound@dialout-78ba,2", "") in new stack == Spawn extension (dialout, outbound, 5) exited non-zero on 'Local/outbound@dialout-78ba,2' Scheduling destruction of SIP dialog '3c0b902d3343200937839d954e419945@proxy2.bandtel.com' in 6400 ms (Method: INVITE) Reliably Transmitting (NAT) to 65.175.129.133:5060: CANCEL sip:011919960466622@proxy2.bandtel.com SIP/2.0 Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK658d6d72;rport From: "2068200001" ;tag=as6de88472 To: Call-ID: 3c0b902d3343200937839d954e419945@proxy2.bandtel.com CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "2068200001" ;privacy=off;screen=no Content-Length: 0 --- Scheduling destruction of SIP dialog '3c0b902d3343200937839d954e419945@proxy2.bandtel.com' in 6400 ms (Method: INVITE) == Spawn extension (dialout, outbound-handler, 1) exited non-zero on 'Local/outbound@dialout-78ba,1' [Dec 4 03:57:24] NOTICE[30721]: pbx_spool.c:351 attempt_thread: Call completed to Local/outbound@dialout Retransmitting #1 (NAT) to 65.175.129.133:5060: CANCEL sip:011919960466622@proxy2.bandtel.com SIP/2.0 Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK658d6d72;rport From: "2068200001" ;tag=as6de88472 To: Call-ID: 3c0b902d3343200937839d954e419945@proxy2.bandtel.com CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "2068200001" ;privacy=off;screen=no Content-Length: 0 --- Retransmitting #2 (NAT) to 65.175.129.133:5060: CANCEL sip:011919960466622@proxy2.bandtel.com SIP/2.0 Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK658d6d72;rport From: "2068200001" ;tag=as6de88472 To: Call-ID: 3c0b902d3343200937839d954e419945@proxy2.bandtel.com CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "2068200001" ;privacy=off;screen=no Content-Length: 0 --- Retransmitting #3 (NAT) to 65.175.129.133:5060: CANCEL sip:011919960466622@proxy2.bandtel.com SIP/2.0 Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK658d6d72;rport From: "2068200001" ;tag=as6de88472 To: Call-ID: 3c0b902d3343200937839d954e419945@proxy2.bandtel.com CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "2068200001" ;privacy=off;screen=no Content-Length: 0 --- Retransmitting #4 (NAT) to 65.175.129.133:5060: CANCEL sip:011919960466622@proxy2.bandtel.com SIP/2.0 Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK658d6d72;rport From: "2068200001" ;tag=as6de88472 To: Call-ID: 3c0b902d3343200937839d954e419945@proxy2.bandtel.com CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "2068200001" ;privacy=off;screen=no Content-Length: 0 --- Retransmitting #5 (NAT) to 65.175.129.133:5060: CANCEL sip:011919960466622@proxy2.bandtel.com SIP/2.0 Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK658d6d72;rport From: "2068200001" ;tag=as6de88472 To: Call-ID: 3c0b902d3343200937839d954e419945@proxy2.bandtel.com CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "2068200001" ;privacy=off;screen=no Content-Length: 0 --- Retransmitting #6 (NAT) to 65.175.129.133:5060: CANCEL sip:011919960466622@proxy2.bandtel.com SIP/2.0 Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK658d6d72;rport From: "2068200001" ;tag=as6de88472 To: Call-ID: 3c0b902d3343200937839d954e419945@proxy2.bandtel.com CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "2068200001" ;privacy=off;screen=no Content-Length: 0 --- Really destroying SIP dialog '3c0b902d3343200937839d954e419945@proxy2.bandtel.com' Method: INVITE <--- SIP read from 65.175.129.133:5060 ---> BYE sip:2068200001@10.1.1.68 SIP/2.0 Via: SIP/2.0/UDP 65.175.129.133;branch=z9hG4bKf68c.1386723063d6adb5111ac6fbb4807938.0 Via: SIP/2.0/UDP 65.175.129.133:5061;branch=z9hG4bKbbdef7a200b9da53df8377e11dda6d13;rport=5061 Max-Forwards: 16 From: ;tag=2a3ad5978d87b904137d93d2eef8307b To: 2068200001 ;tag=as6de88472 Call-ID: 3c0b902d3343200937839d954e419945@proxy2.bandtel.com CSeq: 100 BYE Contact: Anonymous Expires: 300 User-Agent: Sippy cisco-GUID: 1139039190-878584165-3706522926-1638668998 h323-conf-id: 1139039190-878584165-3706522926-1638668998 <-------------> --- (13 headers 0 lines) --- <--- Transmitting (no NAT) to 65.175.129.133:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 65.175.129.133;branch=z9hG4bKf68c.1386723063d6adb5111ac6fbb4807938.0;received=65.175.129.133 Via: SIP/2.0/UDP 65.175.129.133:5061;branch=z9hG4bKbbdef7a200b9da53df8377e11dda6d13;rport=5061 From: ;tag=2a3ad5978d87b904137d93d2eef8307b To: 2068200001 ;tag=as6de88472 Call-ID: 3c0b902d3343200937839d954e419945@proxy2.bandtel.com CSeq: 100 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> [Dec 4 03:58:04] NOTICE[30597]: chan_sip.c:7292 sip_reregister: -- Re-registration for 2068200001@registrar.bandtel.com REGISTER 13 headers, 0 lines Reliably Transmitting (no NAT) to 65.175.129.133:5060: REGISTER sip:registrar.bandtel.com SIP/2.0 Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK2ab21b67;rport From: ;tag=as728b665d To: Call-ID: 7acf76d45513c15e437162be24accd15@10.1.1.68 CSeq: 112 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="2068200001", realm="registrar.bandtel.com", algorithm=MD5, uri="sip:registrar.bandtel.com", nonce="47551ebbe34379de594cebaae453f9a285c1227f", response="2df694733d16158b78608764bb35800d", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 --- <--- SIP read from 65.175.129.133:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK2ab21b67;rport=5060 From: ;tag=as728b665d To: ;tag=e266e8809bf60e12ed80013e395353e4-a437 Call-ID: 7acf76d45513c15e437162be24accd15@10.1.1.68 CSeq: 112 REGISTER PortaBilling: currency:USD Contact: ;expires=295 Server: Sip EXpress router (0.9.6 (i386/freebsd)) Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Scheduling destruction of SIP dialog '7acf76d45513c15e437162be24accd15@10.1.1.68' in 32000 ms (Method: REGISTER) [Dec 4 03:58:04] NOTICE[30597]: chan_sip.c:12289 handle_response_register: Outbound Registration: Expiry for registrar.bandtel.com is 295 sec (Scheduling reregistration in 280 s) Reliably Transmitting (NAT) to 65.175.129.133:5060: OPTIONS sip:proxy2.bandtel.com SIP/2.0 Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK799eaac9;rport From: "asterisk" ;tag=as55bc3120 To: Contact: Call-ID: 29d7b41406896bea38a4afa92ba02d26@10.1.1.68 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 04 Dec 2007 09:58:14 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Retransmitting #1 (NAT) to 65.175.129.133:5060: OPTIONS sip:proxy2.bandtel.com SIP/2.0 Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK799eaac9;rport From: "asterisk" ;tag=as55bc3120 To: Contact: Call-ID: 29d7b41406896bea38a4afa92ba02d26@10.1.1.68 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 04 Dec 2007 09:58:14 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- <--- SIP read from 65.175.129.133:5060 ---> SIP/2.0 501 Not Implemented Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK799eaac9;rport=5060 From: asterisk ;tag=as55bc3120 To: Call-ID: 29d7b41406896bea38a4afa92ba02d26@10.1.1.68 CSeq: 102 OPTIONS Server: Sippy <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '29d7b41406896bea38a4afa92