*CLI> sip set debug SIP Debugging enabled *CLI> -- Attempting call on Local/outbound@dialout for outbound-handler@dialout:1 (Retry 1) -- Executing [outbound@dialout:1] Answer("Local/outbound@dialout-1ddd,2", "") in new stack -- Executing [outbound@dialout:2] Wait("Local/outbound@dialout-1ddd,2", "60") in new stack -- Executing [outbound-handler@dialout:1] Dial("Local/outbound@dialout-1ddd,1", "SIP/011919960466622@proxy2.bandtel.com|50|gM(outbound-connect^agi://10.1.1.68/ivr/speak^---+%0Aname%3A+sanchu%0Aid%3A+1%0A^)") in new stack Audio is at 65.175.129.149 port 6014 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 65.175.129.149:5060;branch=z9hG4bK7203a44d;rport From: "Linux" ;tag=as45a74ee9 To: Contact: Call-ID: 520d660141cdbfc012b36a546d0ac153@proxy2.bandtel.com CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "Linux" ;privacy=off;screen=no Date: Fri, 30 Nov 2007 15:24:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 243 v=0 o=root 31524 31524 IN IP4 65.175.129.149 s=session c=IN IP4 65.175.129.149 t=0 0 m=audio 6014 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 65.175.129.149:5060;branch=z9hG4bK7203a44d;rport=5060;received=10.1.1.68 From: "Linux" ;tag=as45a74ee9 To: Call-ID: 520d660141cdbfc012b36a546d0ac153@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 65.175.129.149:5060;received=10.1.1.68;branch=z9hG4bK7203a44d;rport=5060 Record-Route: From: Linux ;tag=as45a74ee9 To: Call-ID: 520d660141cdbfc012b36a546d0ac153@proxy2.bandtel.com CSeq: 102 INVITE Server: Sippy WWW-Authenticate: Digest realm="65.175.129.133",nonce="ed39e09438a6c4c15a9414d97f6e8e9f47502d86" <-------------> --- (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 65.175.129.149:5060;branch=z9hG4bK7203a44d;rport From: "Linux" ;tag=as45a74ee9 To: Contact: Call-ID: 520d660141cdbfc012b36a546d0ac153@proxy2.bandtel.com CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "Linux" ;privacy=off;screen=no Content-Length: 0 --- Audio is at 65.175.129.149 port 6014 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 65.175.129.149:5060;branch=z9hG4bK253f4eeb;rport From: "Linux" ;tag=as45a74ee9 To: Contact: Call-ID: 520d660141cdbfc012b36a546d0ac153@proxy2.bandtel.com CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "Linux" ;privacy=off;screen=no Authorization: Digest username="2068200001", realm="65.175.129.133", algorithm=MD5, uri="sip:011919960466622@proxy2.bandtel.com", nonce="ed39e09438a6c4c15a9414d97f6e8e9f47502d86", response="5c581beb085f8a548cb9f8aa9a5ac14d", opaque="" Date: Fri, 30 Nov 2007 15:24:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 243 v=0 o=root 31524 31525 IN IP4 65.175.129.149 s=session c=IN IP4 65.175.129.149 t=0 0 m=audio 6014 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 65.175.129.149:5060;branch=z9hG4bK253f4eeb;rport=5060;received=10.1.1.68 From: "Linux" ;tag=as45a74ee9 To: Call-ID: 520d660141cdbfc012b36a546d0ac153@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/2.0 180 Ringing Via: SIP/2.0/UDP 65.175.129.149:5060;received=10.1.1.68;branch=z9hG4bK253f4eeb;rport=5060 Record-Route: From: Linux ;tag=as45a74ee9 To: ;tag=33a5edd5165b767f090f3885e869c297 Call-ID: 520d660141cdbfc012b36a546d0ac153@proxy2.bandtel.com CSeq: 103 INVITE Server: Sippy <-------------> --- (8 headers 0 lines) --- -- SIP/proxy2.bandtel.com-08b6d1e8 is ringing Reliably Transmitting (NAT) to 65.175.129.133:5060: OPTIONS sip:proxy2.bandtel.com SIP/2.0 Via: SIP/2.0/UDP 65.175.129.149:5060;branch=z9hG4bK6d8afb19;rport From: "asterisk" ;tag=as531a2000 To: Contact: Call-ID: 549bc31e370e4bc276eb00d707ef7d46@65.175.129.149 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 30 Nov 2007 15:24:22 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 65.175.129.149:5060;received=10.1.1.68;branch=z9hG4bK6d8afb19;rport=5060 From: asterisk ;tag=as531a2000 To: Call-ID: 549bc31e370e4bc276eb00d707ef7d46@65.175.129.149 CSeq: 102 OPTIONS Server: Sippy <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '549bc31e370e4bc276eb00d707ef7d46@65.175.129.149' Method: OPTIONS -- Nobody picked up in 50000 ms Scheduling destruction of SIP dialog '520d660141cdbfc012b36a546d0ac153@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 65.175.129.149:5060;branch=z9hG4bK253f4eeb;rport From: "Linux" ;tag=as45a74ee9 To: Call-ID: 520d660141cdbfc012b36a546d0ac153@proxy2.bandtel.com CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "Linux" ;privacy=off;screen=no Content-Length: 0 --- Scheduling destruction of SIP dialog '520d660141cdbfc012b36a546d0ac153@proxy2.bandtel.com' in 6400 ms (Method: INVITE) -- Executing [outbound-handler@dialout:2] NoOp("Local/outbound@dialout-1ddd,1", "status_busy=NOANSWER") in new stack -- Executing [outbound-handler@dialout:3] GotoIf("Local/outbound@dialout-1ddd,1", "0?105") in new stack -- Executing [outbound-handler@dialout:4] NoOp("Local/outbound@dialout-1ddd,1", "status=NOANSWER| DIALEDTIME=|ANSWEREDTIME=") in new stack -- Executing [outbound-handler@dialout:5] GotoIf("Local/outbound@dialout-1ddd,1", "0?107") in new stack -- Executing [outbound-handler@dialout:6] GotoIf("Local/outbound@dialout-1ddd,1", "0?101") in new stack -- Executing [outbound-handler@dialout:7] Set("Local/outbound@dialout-1ddd,1", "CallInitiate_hashdata=---+%0Aname%3A+sanchu%0Aid%3A+1%0A") in new stack -- Executing [outbound-handler@dialout:8] Goto("Local/outbound@dialout-1ddd,1", "104") in new stack -- Goto (dialout,outbound-handler,104) -- Executing [outbound-handler@dialout:104] Hangup("Local/outbound@dialout-1ddd,1", "") in new stack == Spawn extension (dialout, outbound-handler, 104) exited non-zero on 'Local/outbound@dialout-1ddd,1' == Spawn extension (dialout, outbound, 2) exited non-zero on 'Local/outbound@dialout-1ddd,2' [Nov 30 09:24:50] NOTICE[1338]: pbx_spool.c:351 attempt_thread: Call completed to Local/outbound@dialout <--- SIP read from 65.175.129.133:5060 ---> SIP/2.0 200 canceling Via: SIP/2.0/UDP 65.175.129.149:5060;branch=z9hG4bK253f4eeb;rport=5060;received=10.1.1.68 From: "Linux" ;tag=as45a74ee9 To: ;tag=e266e8809bf60e12ed80013e395353e4-6224 Call-ID: 520d660141cdbfc012b36a546d0ac153@proxy2.bandtel.com CSeq: 103 CANCEL 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/2.0 487 Request Terminated Via: SIP/2.0/UDP 65.175.129.149:5060;received=10.1.1.68;branch=z9hG4bK253f4eeb;rport=5060 Record-Route: From: Linux ;tag=as45a74ee9 To: Call-ID: 520d660141cdbfc012b36a546d0ac153@proxy2.bandtel.com CSeq: 103 INVITE Server: Sippy <-------------> --- (8 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 65.175.129.149:5060;branch=z9hG4bK253f4eeb;rport From: "Linux" ;tag=as45a74ee9 To: Contact: Call-ID: 520d660141cdbfc012b36a546d0ac153@proxy2.bandtel.com CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "Linux" ;privacy=off;screen=no Content-Length: 0 --- Really destroying SIP dialog '520d660141cdbfc012b36a546d0ac153@proxy2.bandtel.com' Method: INVITE [Nov 30 09:24:54] NOTICE[31550]: chan_sip.c:7292 sip_reregister: -- Re-registration for 2068200001@registrar.bandtel.com REGISTER 13 headers, 0 lines Reliably Transmitting (NAT) to 65.175.129.133:5060: REGISTER sip:registrar.bandtel.com SIP/2.0 Via: SIP/2.0/UDP 65.175.129.149:5060;branch=z9hG4bK50946f3f;rport From: ;tag=as6a832b36 To: Call-ID: 632615d126aa78de05435a3a36e2bfc9@10.1.1.68 CSeq: 128 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="2068200001", realm="registrar.bandtel.com", algorithm=MD5, uri="sip:registrar.bandtel.com", nonce="4750137b6683f579d889ea01f42ff88f933473ed", response="359aebfa7c13f0fa9f7454f065be697b", 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 65.175.129.149:5060;branch=z9hG4bK50946f3f;rport=5060;received=10.1.1.68 From: ;tag=as6a832b36 To: ;tag=e266e8809bf60e12ed80013e395353e4-e737 Call-ID: 632615d126aa78de05435a3a36e2bfc9@10.1.1.68 CSeq: 128 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 '632615d126aa78de05435a3a36e2bfc9@10.1.1.68' in 32000 ms (Method: REGISTER) [Nov 30 09:24:54] NOTICE[31550]: chan_sip.c:12289 handle_response_register: Outbound Registration: Expiry for registrar.bandtel.com is 295 sec (Scheduling reregistration in 280 s) *CLI> Reliably Transmitting (NAT) to 65.175.129.133:5060: OPTIONS sip:proxy2.bandtel.com SIP/2.0 Via: SIP/2.0/UDP 65.175.129.149:5060;branch=z9hG4bK2b6d55ec;rport From: "asterisk" ;tag=as3d1b5dd2 To: Contact: Call-ID: 3c848f8e637ba40b3dc6fe9e3dfa5e46@65.175.129.149 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 30 Nov 2007 15:25:22 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 65.175.129.149:5060;received=10.1.1.68;branch=z9hG4bK2b6d55ec;rport=5060 From: asterisk ;tag=as3d1b5dd2 To: Call-ID: 3c848f8e637ba40b3dc6fe9e3dfa5e46@65.175.129.149 CSeq: 102 OPTIONS Server: Sippy <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '3c848f8e637ba40b3dc6fe9e3dfa5e46@65.175.129.149' Method: OPTIONS Really destroying SIP dialog '632615d126aa78de05435a3a36e2bfc9@10.1.1.68' Method: REGISTER Reliably Transmitting (NAT) to 65.175.129.133:5060: OPTIONS sip:proxy2.bandtel.com SIP/2.0 Via: SIP/2.0/UDP 65.175.129.149:5060;branch=z9hG4bK48cd4715;rport From: "asterisk" ;tag=as43ee5569 To: Contact: Call-ID: 5bdbc34e7914b2cd6628422e4a563478@65.175.129.149 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 30 Nov 2007 15:26:22 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 65.175.129.149:5060;branch=z9hG4bK48cd4715;rport From: "asterisk" ;tag=as43ee5569 To: Contact: Call-ID: 5bdbc34e7914b2cd6628422e4a563478@65.175.129.149 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 30 Nov 2007 15:26:22 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 65.175.129.149:5060;received=10.1.1.68;branch=z9hG4bK48cd4715;rport=5060 From: asterisk ;tag=as43ee5569 To: Call-ID: 5bdbc34e7914b2cd6628422e4a563478@65.175.129.149 CSeq: 102 OPTIONS Server: Sippy <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '5bdbc34e7914b2cd6628422e4a563478@65.175.129.149' Method: OPTIONS