-- Executing [08454632504@service_nts_nextgen_v2:35] Dial("Zap/31-1", "SIP/07919554754@magrathea||jM(service-nts-nextgenv2-register-answer)") in new stack Audio is at 212.84.166.47 port 8338 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Reliably Transmitting (no NAT) to 213.166.5.149:5060: INVITE sip:07919554754@sipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 212.84.166.47:5060;branch=z9hG4bK021b1737;rport From: "01276459900" ;tag=as6fa6a8e8 To: Contact: Call-ID: 604754332387fb286ba7b0a817772c9c@sipgw.magrathea.net CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "01276459900" ;privacy=off;screen=no Date: Wed, 28 Apr 2010 11:10:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 232 v=0 o=root 21929 21929 IN IP4 212.84.166.47 s=session c=IN IP4 212.84.166.47 t=0 0 m=audio 8338 RTP/AVP 8 0 3 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 07919554754@magrathea sc-gw4*CLI> <--- SIP read from 213.166.5.149:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 212.84.166.47:5060;branch=z9hG4bK021b1737;rport=5060 From: "01276459900" ;tag=as6fa6a8e8 To: ;tag=886efdaedc2f197a94b521b96837c731.31e6 Call-ID: 604754332387fb286ba7b0a817772c9c@sipgw.magrathea.net CSeq: 102 INVITE Proxy-Authenticate: Digest realm="magrathea.net", nonce="4bd817aa0001709b5332964428b285a8da79f929b0884cae" Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Transmitting (no NAT) to 213.166.5.149:5060: ACK sip:07919554754@sipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 212.84.166.47:5060;branch=z9hG4bK021b1737;rport From: "01276459900" ;tag=as6fa6a8e8 To: ;tag=886efdaedc2f197a94b521b96837c731.31e6 Contact: Call-ID: 604754332387fb286ba7b0a817772c9c@sipgw.magrathea.net CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "01276459900" ;privacy=off;screen=no Content-Length: 0 --- Audio is at 212.84.166.47 port 8338 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Reliably Transmitting (no NAT) to 213.166.5.149:5060: INVITE sip:07919554754@sipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 212.84.166.47:5060;branch=z9hG4bK1b9bdb03;rport From: "01276459900" ;tag=as6fa6a8e8 To: Contact: Call-ID: 604754332387fb286ba7b0a817772c9c@sipgw.magrathea.net CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "01276459900" ;privacy=off;screen=no Proxy-Authorization: Digest username="skycom", realm="magrathea.net", algorithm=MD5, uri="sip:07919554754@sipgw.magrathea.net", nonce="4bd817aa0001709bxxxxxxxxxxxx29b0884cae", response="df57xxxxxxxxxxxxxxxxgh34a4b21f8a3fd0" Date: Wed, 28 Apr 2010 11:10:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 232 v=0 o=root 21929 21930 IN IP4 212.84.166.47 s=session c=IN IP4 212.84.166.47 t=0 0 m=audio 8338 RTP/AVP 8 0 3 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- sc-gw4*CLI> <--- SIP read from 213.166.5.149:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 212.84.166.47:5060;branch=z9hG4bK1b9bdb03;rport=5060 From: "01276459900" ;tag=as6fa6a8e8 To: Call-ID: 604754332387fb286ba7b0a817772c9c@sipgw.magrathea.net CSeq: 103 INVITE Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- sc-gw4*CLI> <--- SIP read from 213.166.5.149:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 212.84.166.47:5060;received=212.84.166.47;branch=z9hG4bK1b9bdb03;rport=5060 Record-Route: From: "01276459900" ;tag=as6fa6a8e8 To: ;tag=v0Kyr0t5ZvNar Call-ID: 604754332387fb286ba7b0a817772c9c@sipgw.magrathea.net CSeq: 103 INVITE Contact: User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Type: application/sdp Content-Length: 125 v=0 o=- 14294838 0 IN IP4 213.166.5.153 s=Cisco SDP 0 c=IN IP4 213.166.5.153 t=0 0 m=audio 35282 RTP/AVP 8 a=ptime:20 <-------------> --- (13 headers 7 lines) --- Found RTP audio format 8 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) [Apr 28 12:10:08] DEBUG[21939]: chan_sip.c:5623 process_sdp: Our T38 capability = (0), peer T38 capability (0), joint T38 capability (0) Peer audio RTP is at port 213.166.5.153:35282 -- SIP/magrathea-00000002 is making progress passing it to Zap/31-1 [Apr 28 12:10:08] DEBUG[22107]: chan_dahdi.c:5547 dahdi_indicate: Received AST_CONTROL_PROGRESS on Zap/31-1 [phone rings for a short time at this point] -- Channel 0/31, span 1 got hangup request, cause 19 [Apr 28 12:10:22] WARNING[22107]: app_dial.c:720 wait_for_answer: Unable to forward voice frame Scheduling destruction of SIP dialog '604754332387fb286ba7b0a817772c9c@sipgw.magrathea.net' in 32000 ms (Method: INVITE) Reliably Transmitting (no NAT) to 213.166.5.149:5060: CANCEL sip:07919554754@sipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 212.84.166.47:5060;branch=z9hG4bK1b9bdb03;rport From: "01276459900" ;tag=as6fa6a8e8 To: Call-ID: 604754332387fb286ba7b0a817772c9c@sipgw.magrathea.net CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "01276459900" ;privacy=off;screen=no Content-Length: 0 --- Scheduling destruction of SIP dialog '604754332387fb286ba7b0a817772c9c@sipgw.magrathea.net' in 32000 ms (Method: INVITE) == Spawn extension (service_nts_nextgen_v2, 08454632504, 35) exited non-zero on 'Zap/31-1' -- Executing [h@service_nts_nextgen_v2:1] NoOp("Zap/31-1", "number") in new stack -- Executing [h@service_nts_nextgen_v2:2] GotoIf("Zap/31-1", "0?10:3") in new stack -- Goto (service_nts_nextgen_v2,h,3) -- Executing [h@service_nts_nextgen_v2:3] GotoIf("Zap/31-1", "0?20:4") in new stack -- Goto (service_nts_nextgen_v2,h,4) -- Executing [h@service_nts_nextgen_v2:4] DeadAGI("Zap/31-1", "service_nts_nextgen|mode=completed|1272453002.5|CANCEL|1272453002.5||SIP:MAGRATHEA|1|07919554754|0|params=") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/service_nts_nextgen sc-gw4*CLI> <--- SIP read from 213.166.5.149:5060 ---> SIP/2.0 200 canceling Via: SIP/2.0/UDP 212.84.166.47:5060;branch=z9hG4bK1b9bdb03;rport=5060 From: "01276459900" ;tag=as6fa6a8e8 To: ;tag=c17c4f9fe5d2d9daaa1f2eba6fc224d1-861d Call-ID: 604754332387fb286ba7b0a817772c9c@sipgw.magrathea.net CSeq: 103 CANCEL Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 213.166.5.149:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 212.84.166.47:5060;received=212.84.166.47;branch=z9hG4bK1b9bdb03;rport=5060 From: "01276459900" ;tag=as6fa6a8e8 To: ;tag=v0Kyr0t5ZvNar Call-ID: 604754332387fb286ba7b0a817772c9c@sipgw.magrathea.net CSeq: 103 INVITE User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Transmitting (no NAT) to 213.166.5.149:5060: ACK sip:07919554754@sipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 212.84.166.47:5060;branch=z9hG4bK1b9bdb03;rport From: "01276459900" ;tag=as6fa6a8e8 To: ;tag=v0Kyr0t5ZvNar Contact: Call-ID: 604754332387fb286ba7b0a817772c9c@sipgw.magrathea.net CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "01276459900" ;privacy=off;screen=no Content-Length: 0 --- Really destroying SIP dialog '604754332387fb286ba7b0a817772c9c@sipgw.magrathea.net' Method: INVITE -- AGI Script service_nts_nextgen completed, returning 0 -- Executing [h@service_nts_nextgen_v2:5] Hangup("Zap/31-1", "") in new stack == Spawn extension (service_nts_nextgen_v2, h, 5) exited non-zero on 'Zap/31-1'