asterisk*CLI> --- (8 headers 0 lines)--- asterisk*CLI> Looking for s in default (domain 192.168.0.10) asterisk*CLI> Transmitting (no NAT) to 192.168.0.103:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.103;rport;branch=z9hG4bKc0a8006700000010449977674d65efbe00004fbf;received=192.168.0.103 From: ;tag=41519955411501524070 To: ;tag=as6da0be45 Call-ID: 227F676E-1DD2-11B2-A0A9-EDAE4F980423@192.168.0.103 CSeq: 8014 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Destroying call '227F676E-1DD2-11B2-A0A9-EDAE4F980423@192.168.0.103' asterisk*CLI> <-- SIP read from XXX.XXX.XXX.XXX(VOIP Server):5060: BYE sip:8778994498@XXX.XXX.XXX.XXX(Asterisk Server) SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX(VOIP Server):5060;branch=z9hG4bK3dd352a8;rport From: ;tag=as4a3aabda To: "PMG" ;tag=as051bec49 Contact: Call-ID: 0dd66ce37e895a6732bd016d723d5257@XXX.XXX.XXX.XXX(Asterisk Server) CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 asterisk*CLI> --- (10 headers 0 lines)--- asterisk*CLI> Sending to XXX.XXX.XXX.XXX(VOIP Server) : 5060 (NAT) asterisk*CLI> Transmitting (NAT) to XXX.XXX.XXX.XXX(VOIP Server):5060: SIP/2.0 200 OK Via: SIP/2.0/UDP XXX.XXX.XXX.XXX(VOIP Server):5060;branch=z9hG4bK3dd352a8;received=XXX.XXX.XXX.XXX(VOIP Server);rport=5060 From: ;tag=as4a3aabda To: "PMG" ;tag=as051bec49 Call-ID: 0dd66ce37e895a6732bd016d723d5257@XXX.XXX.XXX.XXX(Asterisk Server) CSeq: 102 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- asterisk*CLI> == Spawn extension (rca, **90080817914807480034, 6) exited non-zero on 'SIP/tsr002-699c' asterisk*CLI> set_destination: Parsing for address/port to send to asterisk*CLI> set_destination: set destination to 192.168.0.102, port 5060 asterisk*CLI> Reliably Transmitting (no NAT) to 192.168.0.102:5060: BYE sip:tsr002@192.168.0.102:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK5dee13c4;rport From: "fname_lname_10" ;tag=as509756c9 To: ;tag=4151813532565042683 Contact: Call-ID: 0954a3a6090263c06ca685ba2cb730be@192.168.0.10 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- asterisk*CLI> <-- SIP read from 192.168.0.102:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.10:5060;rport=5060;received=192.168.0.10;branch=z9hG4bK5dee13c4 Content-Length: 0 Call-ID: 0954a3a6090263c06ca685ba2cb730be@192.168.0.10 CSeq: 103 BYE From: "fname_lname_10";tag=as509756c9 To: "unknown";tag=4151813532565042683 --- Destroying call '507ffb9b33b4f669472620795c524744@XXX.XXX.XXX.XXX(VOIP Server)' asterisk*CLI> Destroying call '597D789E-1DD2-11B2-9F5A-CB9D698C9C9E@192.168.0.109' asterisk*CLI> <-- SIP read from 192.168.0.102:5060: OPTIONS sip:192.168.0.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.102;rport;branch=z9hG4bKc0a80066000000104499776533e7297f0000001e Content-Length: 0 Call-ID: B9193A8A-1DD1-11B2-8D86-DF3FAA064735@192.168.0.102 CSeq: 11 OPTIONS From: ;tag=4151993217731562104 Max-Forwards: 70 To: --- (8 headers 0 lines)--- asterisk*CLI> Looking for s in default (domain 192.168.0.10) asterisk*CLI> Transmitting (no NAT) to 192.168.0.102:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.102;rport;branch=z9hG4bKc0a80066000000104499776533e7297f0000001e;received=192.168.0.102 From: ;tag=4151993217731562104 To: ;tag=as27d9de9b Call-ID: B9193A8A-1DD1-11B2-8D86-DF3FAA064735@192.168.0.102 CSeq: 11 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- asterisk*CLI> Destroying call 'B9193A8A-1DD1-11B2-8D86-DF3FAA064735@192.168.0.102' asterisk*CLI> <-- SIP read from 192.168.0.102:5060: REGISTER sip:192.168.0.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.102;rport;branch=z9hG4bKc0a8006600000010449977655472823300000020 Content-Length: 0 Contact: Call-ID: A187E66E-1DD1-11B2-8D86-DF3FAA064735@192.168.0.102 CSeq: 3 REGISTER From: ;tag=41519936791451098011 Max-Forwards: 70 To: asterisk*CLI> --- (9 headers 0 lines)--- asterisk*CLI> Using latest REGISTER request as basis request asterisk*CLI> Sending to 192.168.0.102 : 5060 (NAT) asterisk*CLI> Transmitting (no NAT) to 192.168.0.102:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.102;rport;branch=z9hG4bKc0a8006600000010449977655472823300000020;received=192.168.0.102 From: ;tag=41519936791451098011 To: Call-ID: A187E66E-1DD1-11B2-8D86-DF3FAA064735@192.168.0.102 CSeq: 3 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- asterisk*CLI> Transmitting (no NAT) to 192.168.0.102:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.102;rport;branch=z9hG4bKc0a8006600000010449977655472823300000020;received=192.168.0.102 From: ;tag=41519936791451098011 To: ;tag=as5687ef9f Call-ID: A187E66E-1DD1-11B2-8D86-DF3FAA064735@192.168.0.102 CSeq: 3 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 120 Contact: ;expires=120 Date: Wed, 21 Jun 2006 16:44:21 GMT Content-Length: 0 --- asterisk*CLI> Scheduling destruction of call 'A187E66E-1DD1-11B2-8D86-DF3FAA064735@192.168.0.102' in 15000 ms asterisk*CLI> -- Attempting call on SIP/tsr002 for **90080817924807480034@rca:1 (Retry 1) asterisk*CLI> We're at 192.168.0.10 port 18450 asterisk*CLI> Adding codec 0x2 (gsm) to SDP asterisk*CLI> Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP asterisk*CLI> 13 headers, 11 lines asterisk*CLI> Reliably Transmitting (no NAT) to 192.168.0.102:5060: INVITE sip:tsr002@192.168.0.102:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK38ed3e5b;rport From: "fname_lname_10" ;tag=as1861d1de To: Contact: Call-ID: 34dca93c3d224c2461eb737f29d6a56f@192.168.0.10 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 21 Jun 2006 16:44:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 237 v=0 o=root 5521 5521 IN IP4 192.168.0.10 s=session c=IN IP4 192.168.0.10 t=0 0 m=audio 18450 RTP/AVP 3 0 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- asterisk*CLI> <-- SIP read from 192.168.0.102:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.10:5060;rport=5060;received=192.168.0.10;branch=z9hG4bK38ed3e5b Content-Length: 0 Call-ID: 34dca93c3d224c2461eb737f29d6a56f@192.168.0.10 CSeq: 102 INVITE From: "fname_lname_10";tag=as1861d1de To: "unknown";tag=41519938851223188767 asterisk*CLI> --- (7 headers 0 lines)--- asterisk*CLI> <-- SIP read from 192.168.0.102:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.10:5060;rport=5060;received=192.168.0.10;branch=z9hG4bK38ed3e5b Content-Length: 0 Contact: Call-ID: 34dca93c3d224c2461eb737f29d6a56f@192.168.0.10 CSeq: 102 INVITE From: "fname_lname_10";tag=as1861d1de To: "unknown";tag=41519938851223188767 --- (8 headers 0 lines)--- asterisk*CLI> <-- SIP read from 192.168.0.183:62832: OPTIONS sip:192.168.0.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.100;rport;branch=z9hG4bKc0a86464000000104499775f0000780e00000279 Content-Length: 0 Call-ID: 0488B243-9367-41F7-B989-E169C1AE9B46@192.168.100.102 CSeq: 476 OPTIONS From: ;tag=4333353113333 Max-Forwards: 70 To: --- (8 headers 0 lines)--- Looking for s in default (domain 192.168.0.10) Transmitting (no NAT) to 192.168.0.183:62832: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.100.100;rport;branch=z9hG4bKc0a86464000000104499775f0000780e00000279;received=192.168.0.183 From: ;tag=4333353113333 To: ;tag=as0efb349a Call-ID: 0488B243-9367-41F7-B989-E169C1AE9B46@192.168.100.102 CSeq: 476 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Destroying call '0488B243-9367-41F7-B989-E169C1AE9B46@192.168.100.102' asterisk*CLI> <-- SIP read from 192.168.0.102:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.10:5060;rport=5060;received=192.168.0.10;branch=z9hG4bK38ed3e5b Content-Length: 220 Contact: Call-ID: 34dca93c3d224c2461eb737f29d6a56f@192.168.0.10 Content-Type: application/sdp CSeq: 102 INVITE From: "fname_lname_10";tag=as1861d1de To: "unknown";tag=41519938851223188767 v=0 o=- 3359897061 3359897061 IN IP4 192.168.0.102 s=SJphone c=IN IP4 192.168.0.102 t=0 0 a=direction:active m=audio 49154 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 asterisk*CLI> --- (9 headers 10 lines)--- asterisk*CLI> Found RTP audio format 3 asterisk*CLI> Found RTP audio format 101 asterisk*CLI> Peer audio RTP is at port 192.168.0.102:49154 asterisk*CLI> Found description format GSM Found description format telephone-event asterisk*CLI> Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x2 (gsm)/video=0x0 (nothing), combined - 0x2 (gsm) asterisk*CLI> Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) asterisk*CLI> list_route: hop: asterisk*CLI> set_destination: Parsing for address/port to send to asterisk*CLI> set_destination: set destination to 192.168.0.102, port 5060 asterisk*CLI> Transmitting (no NAT) to 192.168.0.102:5060: ACK sip:tsr002@192.168.0.102:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK08184819;rport From: "fname_lname_10" ;tag=as1861d1de To: ;tag=41519938851223188767 Contact: Call-ID: 34dca93c3d224c2461eb737f29d6a56f@192.168.0.10 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- asterisk*CLI> -- Executing SetCIDNum("SIP/tsr002-01ba", "8778994498") in new stack -- Executing SetCIDName("SIP/tsr002-01ba", "PMG") in new stack -- Executing SetVar("SIP/tsr002-01ba", "CALLFILENAME=9008081792") in new stack -- Executing SetVar("SIP/tsr002-01ba", "CALLPATH=/var/calls") in new stack -- Executing Monitor("SIP/tsr002-01ba", "wav|/var/calls/9008081792|mb") in new stack -- Executing Dial("SIP/tsr002-01ba", "SIP/14807480034@cvcdial|999|r") in new stack We're at XXX.XXX.XXX.XXX(Asterisk Server) port 19578 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 11 lines Reliably Transmitting (no NAT) to XXX.XXX.XXX.XXX(VOIP Server):5060: INVITE sip:14807480034@XXX.XXX.XXX.XXX(VOIP Server) SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX(Asterisk Server):5060;branch=z9hG4bK0deb0424;rport From: "PMG" ;tag=as66fe032e To: Contact: Call-ID: 5a9d853967f57abe0ebbf71b3d34d9ec@XXX.XXX.XXX.XXX(Asterisk Server) CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 21 Jun 2006 16:44:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 235 v=0 o=root 5521 5521 IN IP4 XXX.XXX.XXX.XXX(Asterisk Server) s=session c=IN IP4 XXX.XXX.XXX.XXX(Asterisk Server) t=0 0 m=audio 19578 RTP/AVP 3 0 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called 14807480034@cvcdial asterisk*CLI> <-- SIP read from XXX.XXX.XXX.XXX(VOIP Server):5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP XXX.XXX.XXX.XXX(Asterisk Server):5060;branch=z9hG4bK0deb0424;received=XXX.XXX.XXX.XXX(Asterisk Server);rport=5060 From: "PMG" ;tag=as66fe032e To: Call-ID: 5a9d853967f57abe0ebbf71b3d34d9ec@XXX.XXX.XXX.XXX(Asterisk Server) CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- (10 headers 0 lines)--- asterisk*CLI> <-- SIP read from XXX.XXX.XXX.XXX(VOIP Server):5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP XXX.XXX.XXX.XXX(Asterisk Server):5060;branch=z9hG4bK0deb0424;received=XXX.XXX.XXX.XXX(Asterisk Server);rport=5060 From: "PMG" ;tag=as66fe032e To: ;tag=as1546cb93 Call-ID: 5a9d853967f57abe0ebbf71b3d34d9ec@XXX.XXX.XXX.XXX(Asterisk Server) CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 241 v=0 o=root 23728 23728 IN IP4 XXX.XXX.XXX.XXX(VOIP Server) s=session c=IN IP4 XXX.XXX.XXX.XXX(VOIP Server) t=0 0 m=audio 16950 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - asterisk*CLI> --- (12 headers 0 lines)--- asterisk*CLI> Scheduling destruction of call '3cbee32f1d4f45f158ec08635cfb6a2a@127.0.0.1' in 32000 ms asterisk*CLI> Jun 21 09:44:31 NOTICE[5788]: chan_sip.c:9830 handle_response_register: Outbound Registration: Expiry for 66X235X234X150 is 120 sec (Scheduling reregistration in 105 s) asterisk*CLI> <-- SIP read from XXX.XXX.XXX.XXX(VOIP Server):5060: SIP/2.0 200 OK Via: SIP/2.0/UDP XXX.XXX.XXX.XXX(Asterisk Server):5060;branch=z9hG4bK0deb0424;received=XXX.XXX.XXX.XXX(Asterisk Server);rport=5060 From: "PMG" ;tag=as66fe032e To: ;tag=as1546cb93 Call-ID: 5a9d853967f57abe0ebbf71b3d34d9ec@XXX.XXX.XXX.XXX(Asterisk Server) CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 241 v=0 o=root 23728 23729 IN IP4 XXX.XXX.XXX.XXX(VOIP Server) s=session c=IN IP4 XXX.XXX.XXX.XXX(VOIP Server) t=0 0 m=audio 16950 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - asterisk*CLI> --- (11 headers 11 lines)--- asterisk*CLI> Found RTP audio format 0 Found RTP audio format 3 asterisk*CLI> Found RTP audio format 101 asterisk*CLI> Peer audio RTP is at port XXX.XXX.XXX.XXX(VOIP Server):16950 asterisk*CLI> Found description format PCMU asterisk*CLI> Found description format GSM Found description format telephone-event asterisk*CLI> Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x6 (gsm|ulaw)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw) asterisk*CLI> Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) asterisk*CLI> list_route: hop: asterisk*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to XXX.XXX.XXX.XXX(VOIP Server), port 5060 Transmitting (no NAT) to XXX.XXX.XXX.XXX(VOIP Server):5060: ACK sip:14807480034@XXX.XXX.XXX.XXX(VOIP Server) SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX(Asterisk Server):5060;branch=z9hG4bK6aca22b1;rport From: "PMG" ;tag=as66fe032e To: ;tag=as1546cb93 Contact: Call-ID: 5a9d853967f57abe0ebbf71b3d34d9ec@XXX.XXX.XXX.XXX(Asterisk Server) CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- asterisk*CLI> Destroying call '70CFB4BC-1DD2-11B2-9F5A-CB9D698C9C9E@192.168.0.109' asterisk*CLI> Destroying call 'A187E66E-1DD1-11B2-8D86-DF3FAA064735@192.168.0.102' asterisk*CLI> <-- SIP read from 192.168.0.102:5060: OPTIONS sip:192.168.0.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.102;rport;branch=z9hG4bKc0a80066000000104499777937492daf00000025 Content-Length: 0 Call-ID: B9193A8A-1DD1-11B2-8D86-DF3FAA064735@192.168.0.102 CSeq: 12 OPTIONS From: ;tag=41520132181645676057 Max-Forwards: 70 To: --- (8 headers 0 lines)--- Looking for s in default (domain 192.168.0.10) Transmitting (no NAT) to 192.168.0.102:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.102;rport;branch=z9hG4bKc0a80066000000104499777937492daf00000025;received=192.168.0.102 From: ;tag=41520132181645676057 To: ;tag=as282ea764 Call-ID: B9193A8A-1DD1-11B2-8D86-DF3FAA064735@192.168.0.102 CSeq: 12 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Destroying call '0488B243-9367-41F7-B989-E169C1AE9B46@192.168.100.102' asterisk*CLI> <-- SIP read from XXX.XXX.XXX.XXX(VOIP Server):5060: BYE sip:8778994498@XXX.XXX.XXX.XXX(Asterisk Server) SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX(VOIP Server):5060;branch=z9hG4bK5ded3311;rport From: ;tag=as1546cb93 To: "PMG" ;tag=as66fe032e Contact: Call-ID: 5a9d853967f57abe0ebbf71b3d34d9ec@XXX.XXX.XXX.XXX(Asterisk Server) CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 asterisk*CLI> --- (10 headers 0 lines)--- asterisk*CLI> Sending to XXX.XXX.XXX.XXX(VOIP Server) : 5060 (NAT) asterisk*CLI> Transmitting (NAT) to XXX.XXX.XXX.XXX(VOIP Server):5060: SIP/2.0 200 OK Via: SIP/2.0/UDP XXX.XXX.XXX.XXX(VOIP Server):5060;branch=z9hG4bK5ded3311;received=XXX.XXX.XXX.XXX(VOIP Server);rport=5060 From: ;tag=as1546cb93 To: "PMG" ;tag=as66fe032e Call-ID: 5a9d853967f57abe0ebbf71b3d34d9ec@XXX.XXX.XXX.XXX(Asterisk Server) CSeq: 102 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- asterisk*CLI> == Spawn extension (rca, **90080817924807480034, 6) exited non-zero on 'SIP/tsr002-01ba' asterisk*CLI> set_destination: Parsing for address/port to send to asterisk*CLI> set_destination: set destination to 192.168.0.102, port 5060 asterisk*CLI> Reliably Transmitting (no NAT) to 192.168.0.102:5060: BYE sip:tsr002@192.168.0.102:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK4446f5a4;rport From: "fname_lname_10" ;tag=as1861d1de To: ;tag=41519938851223188767 Contact: Call-ID: 34dca93c3d224c2461eb737f29d6a56f@192.168.0.10 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- asterisk*CLI> <-- SIP read from 192.168.0.102:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.10:5060;rport=5060;received=192.168.0.10;branch=z9hG4bK4446f5a4 Content-Length: 0 Call-ID: 34dca93c3d224c2461eb737f29d6a56f@192.168.0.10 CSeq: 103 BYE From: "fname_lname_10";tag=as1861d1de To: "unknown";tag=41519938851223188767 --- Scheduling destruction of call '05528224-8EA7-4FB0-8DFA-3886115CB2EC@192.168.100.102' in 15000 ms asterisk*CLI> -- Attempting call on SIP/tsr002 for **90080817934807480034@rca:1 (Retry 1) asterisk*CLI> We're at 192.168.0.10 port 19212 asterisk*CLI> Adding codec 0x2 (gsm) to SDP asterisk*CLI> Adding codec 0x4 (ulaw) to SDP asterisk*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk*CLI> 13 headers, 11 lines asterisk*CLI> Reliably Transmitting (no NAT) to 192.168.0.102:5060: INVITE sip:tsr002@192.168.0.102:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK74a79ecd;rport From: "fname_lname_10" ;tag=as2393f69f To: Contact: Call-ID: 48501c4957c98ac50fd4bf573052cff0@192.168.0.10 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 21 Jun 2006 16:44:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 237 v=0 o=root 5521 5521 IN IP4 192.168.0.10 s=session c=IN IP4 192.168.0.10 t=0 0 m=audio 19212 RTP/AVP 3 0 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- asterisk*CLI> <-- SIP read from 192.168.0.102:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.10:5060;rport=5060;received=192.168.0.10;branch=z9hG4bK74a79ecd Content-Length: 0 Call-ID: 48501c4957c98ac50fd4bf573052cff0@192.168.0.10 CSeq: 102 INVITE From: "fname_lname_10";tag=as2393f69f To: "unknown";tag=4152020943107930119 --- (7 headers 0 lines)--- asterisk*CLI> <-- SIP read from 192.168.0.102:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.10:5060;rport=5060;received=192.168.0.10;branch=z9hG4bK74a79ecd Content-Length: 0 Contact: Call-ID: 48501c4957c98ac50fd4bf573052cff0@192.168.0.10 CSeq: 102 INVITE From: "fname_lname_10";tag=as2393f69f To: "unknown";tag=4152020943107930119 --- asterisk*CLI> Transmitting (no NAT) to 192.168.100.100:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.100;rport;branch=z9hG4bKc0a86464000000104499777900006c8a0000027d;received=192.168.0.183 From: ;tag=4335912516060 To: ;tag=as765ba6a4 Call-ID: 05528224-8EA7-4FB0-8DFA-3886115CB2EC@192.168.100.102 CSeq: 129 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 120 Contact: ;expires=120 Date: Wed, 21 Jun 2006 16:44:49 GMT Content-Length: 0 --- asterisk*CLI> Scheduling destruction of call '05528224-8EA7-4FB0-8DFA-3886115CB2EC@192.168.100.102' in 15000 ms asterisk*CLI> <-- SIP read from 192.168.0.102:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.10:5060;rport=5060;received=192.168.0.10;branch=z9hG4bK74a79ecd Content-Length: 220 Contact: Call-ID: 48501c4957c98ac50fd4bf573052cff0@192.168.0.10 Content-Type: application/sdp CSeq: 102 INVITE From: "fname_lname_10";tag=as2393f69f To: "unknown";tag=4152020943107930119 v=0 o=- 3359897088 3359897088 IN IP4 192.168.0.102 s=SJphone c=IN IP4 192.168.0.102 t=0 0 a=direction:active m=audio 49156 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 asterisk*CLI> --- (9 headers 10 lines)--- asterisk*CLI> Found RTP audio format 3 asterisk*CLI> Found RTP audio format 101 asterisk*CLI> Peer audio RTP is at port 192.168.0.102:49156 asterisk*CLI> Found description format GSM asterisk*CLI> Found description format telephone-event asterisk*CLI> Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x2 (gsm)/video=0x0 (nothing), combined - 0x2 (gsm) asterisk*CLI> Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) asterisk*CLI> list_route: hop: asterisk*CLI> set_destination: Parsing for address/port to send to asterisk*CLI> set_destination: set destination to 192.168.0.102, port 5060 asterisk*CLI> Transmitting (no NAT) to 192.168.0.102:5060: ACK sip:tsr002@192.168.0.102:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK111b232f;rport From: "fname_lname_10" ;tag=as2393f69f To: ;tag=4152020943107930119 Contact: Call-ID: 48501c4957c98ac50fd4bf573052cff0@192.168.0.10 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- asterisk*CLI> -- Executing SetCIDNum("SIP/tsr002-1986", "8778994498") in new stack -- Executing SetCIDName("SIP/tsr002-1986", "PMG") in new stack asterisk*CLI> -- Executing SetVar("SIP/tsr002-1986", "CALLFILENAME=9008081793") in new stack asterisk*CLI> -- Executing SetVar("SIP/tsr002-1986", "CALLPATH=/var/calls") in new stack asterisk*CLI> -- Executing Monitor("SIP/tsr002-1986", "wav|/var/calls/9008081793|mb") in new stack asterisk*CLI> -- Executing Dial("SIP/tsr002-1986", "SIP/14807480034@cvcdial|999|r") in new stack asterisk*CLI> We're at XXX.XXX.XXX.XXX(Asterisk Server) port 14492 asterisk*CLI> Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP asterisk*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk*CLI> 13 headers, 11 lines asterisk*CLI> Reliably Transmitting (no NAT) to XXX.XXX.XXX.XXX(VOIP Server):5060: INVITE sip:14807480034@XXX.XXX.XXX.XXX(VOIP Server) SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX(Asterisk Server):5060;branch=z9hG4bK3a784e8f;rport From: "PMG" ;tag=as57bbb343 To: Contact: Call-ID: 4c41d794333fdac3750b768917fc8790@XXX.XXX.XXX.XXX(Asterisk Server) CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 21 Jun 2006 16:44:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 235 v=0 o=root 5521 5521 IN IP4 XXX.XXX.XXX.XXX(Asterisk Server) s=session c=IN IP4 XXX.XXX.XXX.XXX(Asterisk Server) t=0 0 m=audio 14492 RTP/AVP 3 0 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- asterisk*CLI> -- Called 14807480034@cvcdial asterisk*CLI> <-- SIP read from XXX.XXX.XXX.XXX(VOIP Server):5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP XXX.XXX.XXX.XXX(Asterisk Server):5060;branch=z9hG4bK3a784e8f;received=XXX.XXX.XXX.XXX(Asterisk Server);rport=5060 From: "PMG" ;tag=as57bbb343 To: Call-ID: 4c41d794333fdac3750b768917fc8790@XXX.XXX.XXX.XXX(Asterisk Server) CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 asterisk*CLI> --- (10 headers 0 lines)--- asterisk*CLI> <-- SIP read from XXX.XXX.XXX.XXX(VOIP Server):5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP XXX.XXX.XXX.XXX(Asterisk Server):5060;branch=z9hG4bK3a784e8f;received=XXX.XXX.XXX.XXX(Asterisk Server);rport=5060 From: "PMG" ;tag=as57bbb343 To: ;tag=as09aba7a7 Call-ID: 4c41d794333fdac3750b768917fc8790@XXX.XXX.XXX.XXX(Asterisk Server) CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 241 v=0 o=root 23728 23728 IN IP4 XXX.XXX.XXX.XXX(VOIP Server) s=session c=IN IP4 XXX.XXX.XXX.XXX(VOIP Server) t=0 0 m=audio 19026 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- asterisk*CLI> Destroying call '70CFB4BC-1DD2-11B2-9F5A-CB9D698C9C9E@192.168.0.109' asterisk*CLI> <-- SIP read from XXX.XXX.XXX.XXX(VOIP Server):5060: SIP/2.0 200 OK Via: SIP/2.0/UDP XXX.XXX.XXX.XXX(Asterisk Server):5060;branch=z9hG4bK3a784e8f;received=XXX.XXX.XXX.XXX(Asterisk Server);rport=5060 From: "PMG" ;tag=as57bbb343 To: ;tag=as09aba7a7 Call-ID: 4c41d794333fdac3750b768917fc8790@XXX.XXX.XXX.XXX(Asterisk Server) CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 241 v=0 o=root 23728 23729 IN IP4 XXX.XXX.XXX.XXX(VOIP Server) s=session c=IN IP4 XXX.XXX.XXX.XXX(VOIP Server) t=0 0 m=audio 19026 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (11 headers 11 lines)--- asterisk*CLI> Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port XXX.XXX.XXX.XXX(VOIP Server):19026 Found description format PCMU asterisk*CLI> Found description format GSM Found description format telephone-event asterisk*CLI> Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x6 (gsm|ulaw)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: asterisk*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to XXX.XXX.XXX.XXX(VOIP Server), port 5060 asterisk*CLI> Transmitting (no NAT) to XXX.XXX.XXX.XXX(VOIP Server):5060: ACK sip:14807480034@XXX.XXX.XXX.XXX(VOIP Server) SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX(Asterisk Server):5060;branch=z9hG4bK60abb468;rport From: "PMG" ;tag=as57bbb343 To: ;tag=as09aba7a7 Contact: Call-ID: 4c41d794333fdac3750b768917fc8790@XXX.XXX.XXX.XXX(Asterisk Server) CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- asterisk*CLI> Scheduling destruction of call '05528224-8EA7-4FB0-8DFA-3886115CB2EC@192.168.100.102' in 15000 ms asterisk*CLI> <-- SIP read from 192.168.0.102:5060: OPTIONS sip:192.168.0.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.102;rport;branch=z9hG4bKc0a80066000000104499778d02bc94950000002a Content-Length: 0 Call-ID: B9193A8A-1DD1-11B2-8D86-DF3FAA064735@192.168.0.102 CSeq: 13 OPTIONS From: ;tag=4152033219174757187 Max-Forwards: 70 To: asterisk*CLI> --- (8 headers 0 lines)--- asterisk*CLI> Looking for s in default (domain 192.168.0.10) asterisk*CLI> Transmitting (no NAT) to 192.168.0.102:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.102;rport;branch=z9hG4bKc0a80066000000104499778d02bc94950000002a;received=192.168.0.102 From: ;tag=4152033219174757187 To: ;tag=as26c4cbee Call-ID: B9193A8A-1DD1-11B2-8D86-DF3FAA064735@192.168.0.102 CSeq: 13 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- asterisk*CLI> Destroying call '2d79d0d712fb28cf350e8b576cde3f69@XXX.XXX.XXX.XXX(VOIP Server)' asterisk*CLI> Destroying call '05528224-8EA7-4FB0-8DFA-3886115CB2EC@192.168.100.102' asterisk*CLI> <-- SIP read from XXX.XXX.XXX.XXX(VOIP Server):5060: BYE sip:8778994498@XXX.XXX.XXX.XXX(Asterisk Server) SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX(VOIP Server):5060;branch=z9hG4bK3fa5a0e2;rport From: ;tag=as09aba7a7 To: "PMG" ;tag=as57bbb343 Contact: Call-ID: 4c41d794333fdac3750b768917fc8790@XXX.XXX.XXX.XXX(Asterisk Server) CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 asterisk*CLI> --- (10 headers 0 lines)--- asterisk*CLI> Sending to XXX.XXX.XXX.XXX(VOIP Server) : 5060 (NAT) asterisk*CLI> Transmitting (NAT) to XXX.XXX.XXX.XXX(VOIP Server):5060: SIP/2.0 200 OK Via: SIP/2.0/UDP XXX.XXX.XXX.XXX(VOIP Server):5060;branch=z9hG4bK3fa5a0e2;received=XXX.XXX.XXX.XXX(VOIP Server);rport=5060 From: ;tag=as09aba7a7 To: "PMG" ;tag=as57bbb343 Call-ID: 4c41d794333fdac3750b768917fc8790@XXX.XXX.XXX.XXX(Asterisk Server) CSeq: 102 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- asterisk*CLI> == Spawn extension (rca, **90080817934807480034, 6) exited non-zero on 'SIP/tsr002-1986' asterisk*CLI> set_destination: Parsing for address/port to send to asterisk*CLI> set_destination: set destination to 192.168.0.102, port 5060 asterisk*CLI> Reliably Transmitting (no NAT) to 192.168.0.102:5060: BYE sip:tsr002@192.168.0.102:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK3f496560;rport From: "fname_lname_10" ;tag=as2393f69f To: ;tag=4152020943107930119 Contact: Call-ID: 48501c4957c98ac50fd4bf573052cff0@192.168.0.10 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- asterisk*CLI> <-- SIP read from 192.168.0.102:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.10:5060;rport=5060;received=192.168.0.10;branch=z9hG4bK3f496560 Content-Length: 0 Call-ID: 48501c4957c98ac50fd4bf573052cff0@192.168.0.10 CSeq: 103 BYE From: "fname_lname_10";tag=as2393f69f To: "unknown";tag=4152020943107930119 --- (7 headers 0 lines)--- Destroying call '4c41d794333fdac3750b768917fc8790@XXX.XXX.XXX.XXX(Asterisk Server)' Destroying call '48501c4957c98ac50fd4bf573052cff0@192.168.0.10' asterisk*CLI> Jun 21 09:45:19 NOTICE[7154]: pbx_spool.c:279 attempt_thread: Call completed to SIP/tsr002 asterisk*CLI> <-- SIP read from 192.168.0.102:5060: OPTIONS sip:192.168.0.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.102;rport;branch=z9hG4bKc0a8006600000010449977a12529baf50000002d Content-Length: 0 Call-ID: B9193A8A-1DD1-11B2-8D86-DF3FAA064735@192.168.0.102 CSeq: 14 OPTIONS From: ;tag=41520532201796165376 Max-Forwards: 70 To: --- (8 headers 0 lines)--- Looking for s in default (domain 192.168.0.10) Transmitting (no NAT) to 192.168.0.102:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.102;rport;branch=z9hG4bKc0a8006600000010449977a12529baf50000002d;received=192.168.0.102 From: ;tag=41520532201796165376 To: ;tag=as334b8b91 Call-ID: B9193A8A-1DD1-11B2-8D86-DF3FAA064735@192.168.0.102 CSeq: 14 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0