[Sep 29 16:40:52] Asterisk 1.6.1.3-rc1, Copyright (C) 1999 - 2009 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= [Sep 29 16:40:52] Connected to Asterisk 1.6.1.3-rc1 currently running on node4 (pid = 32726) node4*CLI> Verbosity is at least 10 Core debug is at least 10 node4*CLI> sip set debug on node4*CLI> SIP Debugging re-enabled node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK50cd435a;rport=5060 From: "C3259-adnpest-2" ;tag=as33f450ad To: ;tag=B0D10334-18FC Date: Tue, 29 Sep 2009 15:40:44 gmt Call-ID: 177f258f5f3d6df423b928d1184b6a23@sipipgw.magrathea.net Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Allow-Events: telephone-event Contact: Record-Route: Content-Type: application/sdp Content-Length: 249 v=0 o=CiscoSystemsSIP-GW-UserAgent 308 1898 IN IP4 87.238.72.149 s=SIP Call c=IN IP4 87.238.72.149 t=0 0 m=audio 18320 RTP/AVP 8 101 c=IN IP4 87.238.72.149 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (14 headers 11 lines) --- list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.148, port 5060 Transmitting (no NAT) to 213.166.5.148:5060: ACK sip:08448007318381@87.238.72.149:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK3bac7b6f;rport Route: Max-Forwards: 70 From: "C3259-adnpest-2" ;tag=as33f450ad To: ;tag=B0D10334-18FC Contact: Call-ID: 177f258f5f3d6df423b928d1184b6a23@sipipgw.magrathea.net CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "C3259-adnpest-2" ;privacy=off;screen=yes Content-Length: 0 --- -- SIP/magrathea-outbound-b6609c58 answered SIP/87.238.72.149-b7b5b8e8 node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INVITE sip:448081208743@92.63.138.97:5060 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK7dd7.decb2a3.0 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bK7dd7.f1dbf9e5.0 Via: SIP/2.0/UDP 213.166.5.134:5060;branch=z9hG4bK576945A16F1 Remote-Party-ID: ;party=calling;screen=yes;privacy=off From: "01489571331" ;tag=5933CC04-268 To: Date: Tue, 29 Sep 2009 15:41:12 gmt Call-ID: 59689BD9-AC4511DE-BE80D4B2-5B8D3061@213.166.5.134 Supported: timer,resource-priority,replaces Min-SE: 1800 User-Agent: MSSGW Allow: INVITE, BYE, CANCEL, ACK CSeq: 101 INVITE Max-Forwards: 13 Timestamp: 1254238872 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 418 v=0 o=CiscoSystemsSIP-GW-UserAgent 1180 1349 IN IP4 213.166.5.134 s=SIP Call c=IN IP4 213.166.5.134 t=0 0 m=audio 18952 RTP/AVP 8 18 4 3 98 0 101 c=IN IP4 213.166.5.134 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:4 G723/8000 a=fmtp:4 bitrate=6.3;annexa=no a=rtpmap:3 GSM/8000 a=rtpmap:98 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (24 headers 17 lines) --- == Using SIP RTP CoS mark 5 Sending to 92.63.138.100 : 5060 (no NAT) Using INVITE request as basis request - 59689BD9-AC4511DE-BE80D4B2-5B8D3061@213.166.5.134 No matching peer for '01489571331' from '92.63.138.100:5060' Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 3 Found RTP audio format 98 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 213.166.5.134:18952 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Got unsupported a:fmtp in SDP offer Found audio description format G723 for ID 4 Got unsupported a:fmtp in SDP offer Found audio description format GSM for ID 3 Found audio description format G726-32 for ID 98 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8 (alaw), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 213.166.5.134:18952 Looking for 448081208743 in incoming_magrathea (domain 92.63.138.97) node4*CLI> list_route: hop: list_route: hop: <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK7dd7.decb2a3.0;received=92.63.138.100 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bK7dd7.f1dbf9e5.0 Via: SIP/2.0/UDP 213.166.5.134:5060;branch=z9hG4bK576945A16F1 Record-Route: Record-Route: From: "01489571331" ;tag=5933CC04-268 To: Call-ID: 59689BD9-AC4511DE-BE80D4B2-5B8D3061@213.166.5.134 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> node4*CLI> -- Executing AGI("SIP/213.166.5.134-b7b65a58", "agi://web0/track.agi?username=provisioning&campaignName=Number-1&campaignId=913&destination=02076033170&recordCall=yes") node4*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERPRES()=allowed) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(accountcode)=provisioning) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(userfield)=913) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERID(name)=Number-1) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(amaflags)=billing) node4*CLI> -- AGI Script agi://web0/track.agi?username=provisioning&campaignName=Number-1&campaignId=913&destination=02076033170&recordCall=yes completed, returning 0 node4*CLI> -- Executing [448081208743@insight-dialout-external:1] GotoIf("SIP/213.166.5.134-b7b65a58", "1?2:5") in new stack node4*CLI> -- Goto (insight-dialout-external,448081208743,2) node4*CLI> -- Executing [448081208743@insight-dialout-external:2] Answer("SIP/213.166.5.134-b7b65a58", "") in new stack node4*CLI> Audio is at 92.63.138.97 port 14188 node4*CLI> Adding codec 0x8 (alaw) to SDP node4*CLI> Adding non-codec 0x1 (telephone-event) to SDP node4*CLI> <--- Reliably Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK7dd7.decb2a3.0;received=92.63.138.100 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bK7dd7.f1dbf9e5.0 Via: SIP/2.0/UDP 213.166.5.134:5060;branch=z9hG4bK576945A16F1 Record-Route: Record-Route: From: "01489571331" ;tag=5933CC04-268 To: ;tag=as79859efe Call-ID: 59689BD9-AC4511DE-BE80D4B2-5B8D3061@213.166.5.134 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 267 v=0 o=root 1858523804 1858523804 IN IP4 92.63.138.97 s=Asterisk PBX 1.6.1.3-rc1 c=IN IP4 92.63.138.97 t=0 0 m=audio 14188 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> ACK sip:448081208743@92.63.138.97:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK7dd7.decb2a3.2 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bK7dd7.f1dbf9e5.2 Via: SIP/2.0/UDP 213.166.5.134:5060;branch=z9hG4bK576945B14B4 From: ;tag=5933CC04-268 To: ;tag=as79859efe Date: Tue, 29 Sep 2009 15:41:12 gmt Call-ID: 59689BD9-AC4511DE-BE80D4B2-5B8D3061@213.166.5.134 Max-Forwards: 13 CSeq: 101 ACK Allow-Events: telephone-event Content-Length: 0 <-------------> --- (13 headers 0 lines) --- node4*CLI> -- Executing [448081208743@insight-dialout-external:3] MixMonitor("SIP/213.166.5.134-b7b65a58", "voip2-1254238872.627.wav") in new stack node4*CLI> == Begin MixMonitor Recording SIP/213.166.5.134-b7b65a58 node4*CLI> -- Executing [448081208743@insight-dialout-external:4] BackGround("SIP/213.166.5.134-b7b65a58", "adinsight-call-recorded") in new stack node4*CLI> -- Playing 'adinsight-call-recorded.ulaw' (language 'en') node4*CLI> -- Executing [448081208743@insight-dialout-external:5] Dial("SIP/213.166.5.134-b7b65a58", "SIP/magrathea-outbound/02076033170,40,Cr") in new stack node4*CLI> == Using SIP RTP CoS mark 5 node4*CLI> Audio is at 92.63.138.97 port 13652 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP node4*CLI> Reliably Transmitting (no NAT) to 213.166.5.148:5060: INVITE sip:02076033170@sipipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK7532a04d;rport Max-Forwards: 70 From: "Number-1" ;tag=as69e6e6ee To: Contact: Call-ID: 5b6759c357656c505984f86d3ec90c06@sipipgw.magrathea.net CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "Number-1" ;privacy=off;screen=yes Date: Tue, 29 Sep 2009 15:41:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 265 v=0 o=root 294116267 294116267 IN IP4 92.63.138.97 s=Asterisk PBX 1.6.1.3-rc1 c=IN IP4 92.63.138.97 t=0 0 m=audio 13652 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called magrathea-outbound/02076033170 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK7532a04d;rport=5060 From: "Number-1" ;tag=as69e6e6ee To: Call-ID: 5b6759c357656c505984f86d3ec90c06@sipipgw.magrathea.net CSeq: 102 INVITE Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK7532a04d;rport=5060 Record-Route: From: "Number-1" ;tag=as69e6e6ee To: ;tag=B69e0v1eS103N Call-ID: 5b6759c357656c505984f86d3ec90c06@sipipgw.magrathea.net CSeq: 102 INVITE Contact: User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Type: application/sdp Content-Length: 181 v=0 o=- 53370371 0 IN IP4 87.238.72.134 s=Cisco SDP 0 c=IN IP4 87.238.72.134 t=0 0 m=audio 31650 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (13 headers 9 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 87.238.72.134:31650 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 87.238.72.134:31650 node4*CLI> -- SIP/magrathea-outbound-0a15f478 is making progress passing it to SIP/213.166.5.134-b7b65a58 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK7532a04d;rport=5060 Record-Route: From: "Number-1" ;tag=as69e6e6ee To: ;tag=B69e0v1eS103N Call-ID: 5b6759c357656c505984f86d3ec90c06@sipipgw.magrathea.net CSeq: 102 INVITE Contact: User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Min-SE: 120 Content-Type: application/sdp Content-Length: 181 v=0 o=- 53370371 0 IN IP4 87.238.72.134 s=Cisco SDP 0 c=IN IP4 87.238.72.134 t=0 0 m=audio 31650 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (14 headers 9 lines) --- list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.148, port 5060 Transmitting (no NAT) to 213.166.5.148:5060: ACK sip:87.238.72.134 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK15103b94;rport Route: Max-Forwards: 70 From: "Number-1" ;tag=as69e6e6ee To: ;tag=B69e0v1eS103N Contact: Call-ID: 5b6759c357656c505984f86d3ec90c06@sipipgw.magrathea.net CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "Number-1" ;privacy=off;screen=yes Content-Length: 0 --- node4*CLI> -- SIP/magrathea-outbound-0a15f478 answered SIP/213.166.5.134-b7b65a58 node4*CLI> Reliably Transmitting (no NAT) to 213.166.5.148:5060: OPTIONS sip:sipipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK3f6dc8bd;rport Max-Forwards: 70 From: "asterisk" ;tag=as73018988 To: Contact: Call-ID: 50e4505b32a39d4d200eb6960323affb@92.63.138.97 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.1.3-rc1 Date: Tue, 29 Sep 2009 15:41:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 403 OPTIONS not supported Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK3f6dc8bd;rport=5060 From: "asterisk" ;tag=as73018988 To: ;tag=9a264c9a00f926193bf7ce80aab147c3.aa7b Call-ID: 50e4505b32a39d4d200eb6960323affb@92.63.138.97 CSeq: 102 OPTIONS Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '50e4505b32a39d4d200eb6960323affb@92.63.138.97' Method: OPTIONS node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INFO sip:92.63.138.97:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKfaae.2851a3e2.0 To: sip:92.63.138.97:5060 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-5e94 CSeq: 10 INFO Call-ID: 6ad64922-7670@92.63.138.100 Content-Length: 0 User-Agent: OpenSIPS (1.5.1-notls (i386/linux)) <-------------> --- (8 headers 0 lines) --- <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKfaae.2851a3e2.0;received=92.63.138.100 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-5e94 To: sip:92.63.138.97:5060;tag=as7eab632a Call-ID: 6ad64922-7670@92.63.138.100 CSeq: 10 INFO Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> node4*CLI> node4*CLI> node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> BYE sip:448005244626@92.63.138.97:5060 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKb326.0c9c7ea2.0 Via: SIP/2.0/UDP 87.238.72.153;branch=z9hG4bKb326.ded49124.0 Via: SIP/2.0/UDP 87.238.72.149:5060;branch=z9hG4bK1694A9D1F50 From: ;tag=B0D09D5C-135 To: ;tag=as3bba9d95 node4*CLI> Date: Tue, 29 Sep 2009 15:40:18 gmt Call-ID: 39B7C6DA-AC4511DE-B574F3EC-65114225@87.238.72.149 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 13 Timestamp: 1254238889 CSeq: 102 BYE Reason: Q.850;cause=16 Content-Length: 0 <-------------> --- (16 headers 0 lines) --- Sending to 92.63.138.100 : 5060 (no NAT) <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKb326.0c9c7ea2.0;received=92.63.138.100 Via: SIP/2.0/UDP 87.238.72.153;branch=z9hG4bKb326.ded49124.0 Via: SIP/2.0/UDP 87.238.72.149:5060;branch=z9hG4bK1694A9D1F50 Record-Route: Record-Route: From: ;tag=B0D09D5C-135 To: ;tag=as3bba9d95 Call-ID: 39B7C6DA-AC4511DE-B574F3EC-65114225@87.238.72.149 CSeq: 102 BYE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> -- Executing [h@insight-dialout-external:1] Set("SIP/87.238.72.149-b7b7fef0", "CDR(outbound)=08456860074") in new stack node4*CLI> Scheduling destruction of SIP dialog '181481b363c16040004d5d4958e669f1@sipipgw.magrathea.net' in 6400 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.148, port 5060 Reliably Transmitting (no NAT) to 213.166.5.148:5060: BYE sip:08448456860074@87.238.72.155:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK61c5908d;rport Route: Max-Forwards: 70 From: "C12808-hipsvault" ;tag=as77b29a0c To: ;tag=A5CA98A8-103C Call-ID: 181481b363c16040004d5d4958e669f1@sipipgw.magrathea.net CSeq: 103 BYE User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "C12808-hipsvault" ;privacy=off;screen=yes X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- == Spawn extension (insight-dialout-external, 448005244626, 5) exited non-zero on 'SIP/87.238.72.149-b7b7fef0' node4*CLI> == MixMonitor close filestream node4*CLI> == End MixMonitor Recording SIP/87.238.72.149-b7b7fef0 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK61c5908d;rport=5060 From: "C12808-hipsvault" ;tag=as77b29a0c To: ;tag=A5CA98A8-103C Date: Tue, 29 Sep 2009 15:41:29 gmt Call-ID: 181481b363c16040004d5d4958e669f1@sipipgw.magrathea.net Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 103 BYE <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '39B7C6DA-AC4511DE-B574F3EC-65114225@87.238.72.149' Method: BYE Really destroying SIP dialog '181481b363c16040004d5d4958e669f1@sipipgw.magrathea.net' Method: INVITE node4*CLI> node4*CLI> node4*CLI> node4*CLI> node4*CLI> node4*CLI> node4*CLI> node4*CLI> node4*CLI> node4*CLI> node4*CLI> node4*CLI> node4*CLI> node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INVITE sip:441616607985@92.63.138.97:5060 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKdd5c.4f8d52d1.0 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bKdd5c.09b1aac6.0 Via: SIP/2.0/UDP 213.166.5.140:5060;branch=z9hG4bK503F9423DF Remote-Party-ID: ;party=calling;screen=yes;privacy=off From: "07971296525" ;tag=C668B120-1FDD To: Date: Tue, 29 Sep 2009 15:41:34 gmt Call-ID: 66B6355D-AC4511DE-9169D31F-EFD16753@213.166.5.140 Supported: timer,resource-priority,replaces Min-SE: 1800 User-Agent: MSSGW Allow: INVITE, BYE, CANCEL, ACK CSeq: 101 INVITE Max-Forwards: 13 Timestamp: 1254238894 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 418 v=0 o=CiscoSystemsSIP-GW-UserAgent 2573 3265 IN IP4 213.166.5.140 s=SIP Call c=IN IP4 213.166.5.140 t=0 0 m=audio 19932 RTP/AVP 8 18 4 3 98 0 101 c=IN IP4 213.166.5.140 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:4 G723/8000 a=fmtp:4 bitrate=6.3;annexa=no a=rtpmap:3 GSM/8000 a=rtpmap:98 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (24 headers 17 lines) --- node4*CLI> == Using SIP RTP CoS mark 5 Sending to 92.63.138.100 : 5060 (no NAT) Using INVITE request as basis request - 66B6355D-AC4511DE-9169D31F-EFD16753@213.166.5.140 No matching peer for '07971296525' from '92.63.138.100:5060' Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 3 Found RTP audio format 98 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 213.166.5.140:19932 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Got unsupported a:fmtp in SDP offer Found audio description format G723 for ID 4 Got unsupported a:fmtp in SDP offer Found audio description format GSM for ID 3 Found audio description format G726-32 for ID 98 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8 (alaw), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 213.166.5.140:19932 Looking for 441616607985 in incoming_magrathea (domain 92.63.138.97) node4*CLI> list_route: hop: list_route: hop: node4*CLI> <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKdd5c.4f8d52d1.0;received=92.63.138.100 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bKdd5c.09b1aac6.0 Via: SIP/2.0/UDP 213.166.5.140:5060;branch=z9hG4bK503F9423DF Record-Route: Record-Route: From: "07971296525" ;tag=C668B120-1FDD To: node4*CLI> Call-ID: 66B6355D-AC4511DE-9169D31F-EFD16753@213.166.5.140 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> node4*CLI> -- Executing AGI("SIP/213.166.5.140-b7b80688", "agi://web0/track.agi?username=rhamnett&campaignName=fault+test&campaignId=2681&destination=07841131297&salesTracking=yes&recordCall=yes") node4*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERPRES()=allowed) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(accountcode)=rhamnett) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(userfield)=2681) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERID(name)=fault test) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(amaflags)=billing) node4*CLI> -- AGI Script agi://web0/track.agi?username=rhamnett&campaignName=fault+test&campaignId=2681&destination=07841131297&salesTracking=yes&recordCall=yes completed, returning 0 node4*CLI> -- Executing [441616607985@insight-dialout-external:1] GotoIf("SIP/213.166.5.140-b7b80688", "1?2:5") in new stack -- Goto (insight-dialout-external,441616607985,2) -- Executing [441616607985@insight-dialout-external:2] Answer("SIP/213.166.5.140-b7b80688", "") in new stack node4*CLI> Audio is at 92.63.138.97 port 11888 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP node4*CLI> <--- Reliably Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKdd5c.4f8d52d1.0;received=92.63.138.100 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bKdd5c.09b1aac6.0 Via: SIP/2.0/UDP 213.166.5.140:5060;branch=z9hG4bK503F9423DF Record-Route: Record-Route: From: "07971296525" ;tag=C668B120-1FDD To: ;tag=as598e3492 Call-ID: 66B6355D-AC4511DE-9169D31F-EFD16753@213.166.5.140 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 267 v=0 o=root 1025224544 1025224544 IN IP4 92.63.138.97 s=Asterisk PBX 1.6.1.3-rc1 c=IN IP4 92.63.138.97 t=0 0 m=audio 11888 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> ACK sip:441616607985@92.63.138.97:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKdd5c.4f8d52d1.2 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bKdd5c.09b1aac6.2 Via: SIP/2.0/UDP 213.166.5.140:5060;branch=z9hG4bK503F95200C From: ;tag=C668B120-1FDD To: ;tag=as598e3492 Date: Tue, 29 Sep 2009 15:41:34 gmt Call-ID: 66B6355D-AC4511DE-9169D31F-EFD16753@213.166.5.140 Max-Forwards: 13 CSeq: 101 ACK Allow-Events: telephone-event Content-Length: 0 <-------------> --- (13 headers 0 lines) --- node4*CLI> -- Executing [441616607985@insight-dialout-external:3] MixMonitor("SIP/213.166.5.140-b7b80688", "voip2-1254238894.629.wav") in new stack node4*CLI> -- Executing [441616607985@insight-dialout-external:4] BackGround("SIP/213.166.5.140-b7b80688", "adinsight-call-recorded") in new stack node4*CLI> == Begin MixMonitor Recording SIP/213.166.5.140-b7b80688 node4*CLI> -- Playing 'adinsight-call-recorded.ulaw' (language 'en') node4*CLI> node4*CLI> node4*CLI> node4*CLI> -- Executing [441616607985@insight-dialout-external:5] Dial("SIP/213.166.5.140-b7b80688", "SIP/magrathea-outbound/07841131297,40,CrF(trackSales^441616607985:2681:voip2-1254238894.629:07841131297^1)") in new stack node4*CLI> == Using SIP RTP CoS mark 5 node4*CLI> Audio is at 92.63.138.97 port 13196 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 213.166.5.148:5060: INVITE sip:07841131297@sipipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK03523bd3;rport Max-Forwards: 70 From: "fault test" ;tag=as44f78ecd To: Contact: Call-ID: 388b46b059088c105173be163249b34e@sipipgw.magrathea.net CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "fault test" ;privacy=off;screen=yes Date: Tue, 29 Sep 2009 15:41:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 267 v=0 o=root 2102960701 2102960701 IN IP4 92.63.138.97 s=Asterisk PBX 1.6.1.3-rc1 c=IN IP4 92.63.138.97 t=0 0 m=audio 13196 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called magrathea-outbound/07841131297 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK03523bd3;rport=5060 From: "fault test" ;tag=as44f78ecd To: Call-ID: 388b46b059088c105173be163249b34e@sipipgw.magrathea.net CSeq: 102 INVITE Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK03523bd3;rport=5060 Record-Route: From: "fault test" ;tag=as44f78ecd To: ;tag=H74SUjgXeQ88r Call-ID: 388b46b059088c105173be163249b34e@sipipgw.magrathea.net CSeq: 102 INVITE Contact: User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Type: application/sdp Content-Length: 181 v=0 o=- 38568868 0 IN IP4 87.238.72.134 s=Cisco SDP 0 c=IN IP4 87.238.72.134 t=0 0 m=audio 32078 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (13 headers 9 lines) --- node4*CLI> Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 87.238.72.134:32078 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 87.238.72.134:32078 -- SIP/magrathea-outbound-b6619730 is making progress passing it to SIP/213.166.5.140-b7b80688 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK03523bd3;rport=5060 Record-Route: From: "fault test" ;tag=as44f78ecd To: ;tag=H74SUjgXeQ88r Call-ID: 388b46b059088c105173be163249b34e@sipipgw.magrathea.net CSeq: 102 INVITE Contact: User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Min-SE: 120 Content-Type: application/sdp Content-Length: 181 v=0 o=- 38568868 0 IN IP4 87.238.72.134 s=Cisco SDP 0 c=IN IP4 87.238.72.134 t=0 0 m=audio 32078 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (14 headers 9 lines) --- node4*CLI> list_route: hop: node4*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.148, port 5060 Transmitting (no NAT) to 213.166.5.148:5060: ACK sip:87.238.72.134 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK21276e48;rport Route: Max-Forwards: 70 From: "fault test" ;tag=as44f78ecd To: ;tag=H74SUjgXeQ88r Contact: Call-ID: 388b46b059088c105173be163249b34e@sipipgw.magrathea.net CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "fault test" ;privacy=off;screen=yes Content-Length: 0 --- node4*CLI> -- SIP/magrathea-outbound-b6619730 answered SIP/213.166.5.140-b7b80688 node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> BYE sip:448081208518@92.63.138.97:5060 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK8953.67e62a63.0 Via: SIP/2.0/UDP 87.238.72.153;branch=z9hG4bK8953.14bec88.0 Via: SIP/2.0/UDP 87.238.72.149:5060;branch=z9hG4bK1694AC4502 From: ;tag=B0D0F514-9B3 To: ;tag=as2b2d8644 Date: Tue, 29 Sep 2009 15:40:41 gmt Call-ID: 471AEB88-AC4511DE-B5DBF3EC-65114225@87.238.72.149 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 13 Timestamp: 1254238909 CSeq: 102 BYE Reason: Q.850;cause=16 Content-Length: 0 <-------------> --- (16 headers 0 lines) --- Sending to 92.63.138.100 : 5060 (no NAT) node4*CLI> <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK8953.67e62a63.0;received=92.63.138.100 Via: SIP/2.0/UDP 87.238.72.153;branch=z9hG4bK8953.14bec88.0 Via: SIP/2.0/UDP 87.238.72.149:5060;branch=z9hG4bK1694AC4502 Record-Route: Record-Route: From: ;tag=B0D0F514-9B3 To: ;tag=as2b2d8644 Call-ID: 471AEB88-AC4511DE-B5DBF3EC-65114225@87.238.72.149 CSeq: 102 BYE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> -- Executing [h@insight-dialout-external:1] Set("SIP/87.238.72.149-b7b5b8e8", "CDR(outbound)=08007318381") in new stack node4*CLI> Scheduling destruction of SIP dialog '177f258f5f3d6df423b928d1184b6a23@sipipgw.magrathea.net' in 6400 ms (Method: INVITE) node4*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.148, port 5060 Reliably Transmitting (no NAT) to 213.166.5.148:5060: BYE sip:08448007318381@87.238.72.149:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK6d01a75a;rport Route: Max-Forwards: 70 From: "C3259-adnpest-2" ;tag=as33f450ad To: ;tag=B0D10334-18FC Call-ID: 177f258f5f3d6df423b928d1184b6a23@sipipgw.magrathea.net CSeq: 103 BYE User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "C3259-adnpest-2" ;privacy=off;screen=yes X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- node4*CLI> == Spawn extension (insight-dialout-external, 448081208518, 5) exited non-zero on 'SIP/87.238.72.149-b7b5b8e8' node4*CLI> == MixMonitor close filestream == End MixMonitor Recording SIP/87.238.72.149-b7b5b8e8 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK6d01a75a;rport=5060 From: "C3259-adnpest-2" ;tag=as33f450ad To: ;tag=B0D10334-18FC Date: Tue, 29 Sep 2009 15:41:49 gmt Call-ID: 177f258f5f3d6df423b928d1184b6a23@sipipgw.magrathea.net Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 103 BYE <-------------> --- (9 headers 0 lines) --- node4*CLI> Really destroying SIP dialog '177f258f5f3d6df423b928d1184b6a23@sipipgw.magrathea.net' Method: INVITE Really destroying SIP dialog '471AEB88-AC4511DE-B5DBF3EC-65114225@87.238.72.149' Method: BYE node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INFO sip:92.63.138.97:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKbaae.a81f47a2.0 To: sip:92.63.138.97:5060 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-e8c2 CSeq: 10 INFO Call-ID: 6ad64926-7670@92.63.138.100 Content-Length: 0 User-Agent: OpenSIPS (1.5.1-notls (i386/linux)) <-------------> --- (8 headers 0 lines) --- node4*CLI> <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKbaae.a81f47a2.0;received=92.63.138.100 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-e8c2 To: sip:92.63.138.97:5060;tag=as40079510 Call-ID: 6ad64926-7670@92.63.138.100 CSeq: 10 INFO Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> BYE sip:448005244845@92.63.138.97:5060 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK22a3.ec5bb833.0 Via: SIP/2.0/UDP 87.238.72.153;branch=z9hG4bK22a3.22e67af7.0 Via: SIP/2.0/UDP 87.238.72.155:5060;branch=z9hG4bK1FCCC42EA From: "anonymous" ;tag=A5C55CD8-D8F To: ;tag=as5927da31 Date: Tue, 29 Sep 2009 15:34:39 gmt Call-ID: 6F6EB10D-AC4411DE-85FDBA8E-F0DAC5A2@87.238.72.155 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 13 Timestamp: 1254238912 CSeq: 102 BYE Reason: Q.850;cause=16 Content-Length: 0 <-------------> --- (16 headers 0 lines) --- Sending to 92.63.138.100 : 5060 (no NAT) node4*CLI> <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK22a3.ec5bb833.0;received=92.63.138.100 Via: SIP/2.0/UDP 87.238.72.153;branch=z9hG4bK22a3.22e67af7.0 Via: SIP/2.0/UDP 87.238.72.155:5060;branch=z9hG4bK1FCCC42EA Record-Route: Record-Route: From: "anonymous" ;tag=A5C55CD8-D8F To: ;tag=as5927da31 Call-ID: 6F6EB10D-AC4411DE-85FDBA8E-F0DAC5A2@87.238.72.155 CSeq: 102 BYE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> -- Executing [h@insight-dialout-external:1] Set("SIP/87.238.72.155-b7b83568", "CDR(outbound)=08450744027") in new stack node4*CLI> Scheduling destruction of SIP dialog '0edf5f3a6b12ec5a7358cc087bd4fec1@sipipgw.magrathea.net' in 6400 ms (Method: INVITE) node4*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.148, port 5060 Reliably Transmitting (no NAT) to 213.166.5.148:5060: BYE sip:08448450744027@87.238.72.155:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK044fa8c9;rport Route: Max-Forwards: 70 From: "Affiliate 1" ;tag=as0d86a1a4 To: ;tag=A5C55D0C-1E13 Call-ID: 0edf5f3a6b12ec5a7358cc087bd4fec1@sipipgw.magrathea.net CSeq: 103 BYE User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "Affiliate 1" ;privacy=off;screen=yes X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- node4*CLI> == Spawn extension (insight-dialout-external, 448005244845, 5) exited non-zero on 'SIP/87.238.72.155-b7b83568' node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK044fa8c9;rport=5060 From: "Affiliate 1" ;tag=as0d86a1a4 To: ;tag=A5C55D0C-1E13 Date: Tue, 29 Sep 2009 15:41:52 gmt Call-ID: 0edf5f3a6b12ec5a7358cc087bd4fec1@sipipgw.magrathea.net Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 103 BYE <-------------> --- (9 headers 0 lines) --- node4*CLI> Really destroying SIP dialog '0edf5f3a6b12ec5a7358cc087bd4fec1@sipipgw.magrathea.net' Method: INVITE Really destroying SIP dialog '6F6EB10D-AC4411DE-85FDBA8E-F0DAC5A2@87.238.72.155' Method: BYE node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INVITE sip:448447041620@92.63.138.97:5060 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK34d4.874b2d92.0 Via: SIP/2.0/UDP 87.238.72.153;branch=z9hG4bK34d4.b72665e3.0 Via: SIP/2.0/UDP 87.238.72.155:5060;branch=z9hG4bK1FCCCD1BCE From: "01834861756" ;tag=A5CC1024-F70 To: Date: Tue, 29 Sep 2009 15:41:58 gmt Call-ID: 752A8F84-AC4511DE-8E6ABA8E-F0DAC5A2@87.238.72.155 Supported: timer Min-SE: 1800 User-Agent: MSSGW(B) Allow: INVITE, BYE, CANCEL, ACK CSeq: 101 INVITE Max-Forwards: 13 Remote-Party-ID: ;party=calling;screen=yes;privacy=off Timestamp: 1254238918 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 404 v=0 o=CiscoSystemsSIP-GW-UserAgent 6844 6166 IN IP4 87.238.72.155 s=SIP Call c=IN IP4 87.238.72.155 t=0 0 m=audio 17658 RTP/AVP 8 18 4 3 2 0 101 c=IN IP4 87.238.72.155 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:3 GSM/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (23 headers 17 lines) --- node4*CLI> == Using SIP RTP CoS mark 5 Sending to 92.63.138.100 : 5060 (no NAT) Using INVITE request as basis request - 752A8F84-AC4511DE-8E6ABA8E-F0DAC5A2@87.238.72.155 No matching peer for '01834861756' from '92.63.138.100:5060' node4*CLI> Found RTP audio format 8 Found RTP audio format 18 node4*CLI> Found RTP audio format 4 Found RTP audio format 3 Found RTP audio format 2 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 87.238.72.155:17658 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Got unsupported a:fmtp in SDP offer Found audio description format G723 for ID 4 Got unsupported a:fmtp in SDP offer Found audio description format GSM for ID 3 Found audio description format G726-32 for ID 2 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8 (alaw), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 87.238.72.155:17658 Looking for 448447041620 in incoming_magrathea (domain 92.63.138.97) node4*CLI> list_route: hop: list_route: hop: node4*CLI> <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK34d4.874b2d92.0;received=92.63.138.100 Via: SIP/2.0/UDP 87.238.72.153;branch=z9hG4bK34d4.b72665e3.0 Via: SIP/2.0/UDP 87.238.72.155:5060;branch=z9hG4bK1FCCCD1BCE Record-Route: Record-Route: From: "01834861756" ;tag=A5CC1024-F70 To: Call-ID: 752A8F84-AC4511DE-8E6ABA8E-F0DAC5A2@87.238.72.155 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> node4*CLI> -- Executing AGI("SIP/87.238.72.155-b7b87118", "agi://web0/track.agi?username=emsinternet&campaignName=Faxes&campaignId=215&destination=01925413333&salesTracking=yes&recordCall=yes&analytics=yes") node4*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERPRES()=allowed) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(accountcode)=emsinternet) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(userfield)=215) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERID(name)=Faxes) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(amaflags)=billing) node4*CLI> -- AGI Script agi://web0/track.agi?username=emsinternet&campaignName=Faxes&campaignId=215&destination=01925413333&salesTracking=yes&recordCall=yes&analytics=yes completed, returning 0 node4*CLI> -- Executing [448447041620@insight-dialout-external:1] GotoIf("SIP/87.238.72.155-b7b87118", "1?2:5") in new stack -- Goto (insight-dialout-external,448447041620,2) -- Executing [448447041620@insight-dialout-external:2] Answer("SIP/87.238.72.155-b7b87118", "") in new stack node4*CLI> Audio is at 92.63.138.97 port 16042 Adding codec 0x8 (alaw) to SDP node4*CLI> Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK34d4.874b2d92.0;received=92.63.138.100 Via: SIP/2.0/UDP 87.238.72.153;branch=z9hG4bK34d4.b72665e3.0 Via: SIP/2.0/UDP 87.238.72.155:5060;branch=z9hG4bK1FCCCD1BCE Record-Route: Record-Route: From: "01834861756" ;tag=A5CC1024-F70 To: ;tag=as7d091253 Call-ID: 752A8F84-AC4511DE-8E6ABA8E-F0DAC5A2@87.238.72.155 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 265 v=0 o=root 643492988 643492988 IN IP4 92.63.138.97 s=Asterisk PBX 1.6.1.3-rc1 c=IN IP4 92.63.138.97 t=0 0 m=audio 16042 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> ACK sip:448447041620@92.63.138.97:5060 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK34d4.874b2d92.2 Via: SIP/2.0/UDP 87.238.72.153;branch=z9hG4bK34d4.b72665e3.2 Via: SIP/2.0/UDP 87.238.72.155:5060;branch=z9hG4bK1FCCCE782 From: ;tag=A5CC1024-F70 To: ;tag=as7d091253 Date: Tue, 29 Sep 2009 15:41:58 gmt Call-ID: 752A8F84-AC4511DE-8E6ABA8E-F0DAC5A2@87.238.72.155 Max-Forwards: 13 CSeq: 101 ACK Content-Length: 0 <-------------> --- (13 headers 0 lines) --- node4*CLI> -- Executing [448447041620@insight-dialout-external:3] MixMonitor("SIP/87.238.72.155-b7b87118", "voip2-1254238918.631.wav") in new stack node4*CLI> -- Executing [448447041620@insight-dialout-external:4] BackGround("SIP/87.238.72.155-b7b87118", "adinsight-call-recorded") in new stack node4*CLI> == Begin MixMonitor Recording SIP/87.238.72.155-b7b87118 node4*CLI> -- Playing 'adinsight-call-recorded.ulaw' (language 'en') node4*CLI> node4*CLI> node4*CLI> node4*CLI> node4*CLI> -- Executing [448447041620@insight-dialout-external:5] Dial("SIP/87.238.72.155-b7b87118", "SIP/magrathea-outbound/01925413333,40,CrF(trackSales^448447041620:215:voip2-1254238918.631:01925413333^1)") in new stack node4*CLI> == Using SIP RTP CoS mark 5 node4*CLI> Audio is at 92.63.138.97 port 10860 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP node4*CLI> Reliably Transmitting (no NAT) to 213.166.5.148:5060: INVITE sip:01925413333@sipipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK685a4fa3;rport Max-Forwards: 70 From: "Faxes" ;tag=as0980f15f To: Contact: Call-ID: 6cac8ce45b7ae4644ba7056b742799c7@sipipgw.magrathea.net CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "Faxes" ;privacy=off;screen=yes Date: Tue, 29 Sep 2009 15:42:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 267 v=0 o=root 1127971354 1127971354 IN IP4 92.63.138.97 s=Asterisk PBX 1.6.1.3-rc1 c=IN IP4 92.63.138.97 t=0 0 m=audio 10860 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called magrathea-outbound/01925413333 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK685a4fa3;rport=5060 From: "Faxes" ;tag=as0980f15f To: Call-ID: 6cac8ce45b7ae4644ba7056b742799c7@sipipgw.magrathea.net CSeq: 102 INVITE Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- node4*CLI> node4*CLI> node4*CLI> node4*CLI> node4*CLI> node4*CLI> node4*CLI> node4*CLI> node4*CLI> node4*CLI> node4*CLI> node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK685a4fa3;rport=5060 Record-Route: From: "Faxes" ;tag=as0980f15f To: ;tag=aetyyy3p04p0F Call-ID: 6cac8ce45b7ae4644ba7056b742799c7@sipipgw.magrathea.net CSeq: 102 INVITE Contact: User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Type: application/sdp Content-Length: 181 v=0 o=- 38569147 0 IN IP4 213.166.5.154 s=Cisco SDP 0 c=IN IP4 213.166.5.154 t=0 0 m=audio 34246 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (13 headers 9 lines) --- node4*CLI> Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 213.166.5.154:34246 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) node4*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 213.166.5.154:34246 -- SIP/magrathea-outbound-0a1bd258 is making progress passing it to SIP/87.238.72.155-b7b87118 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK685a4fa3;rport=5060 Record-Route: From: "Faxes" ;tag=as0980f15f To: ;tag=aetyyy3p04p0F Call-ID: 6cac8ce45b7ae4644ba7056b742799c7@sipipgw.magrathea.net CSeq: 102 INVITE Contact: User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Min-SE: 120 Content-Type: application/sdp Content-Length: 181 v=0 o=- 38569147 0 IN IP4 213.166.5.154 s=Cisco SDP 0 c=IN IP4 213.166.5.154 t=0 0 m=audio 34246 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (14 headers 9 lines) --- node4*CLI> list_route: hop: node4*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.148, port 5060 node4*CLI> Transmitting (no NAT) to 213.166.5.148:5060: ACK sip:213.166.5.154 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK23ae5512;rport Route: Max-Forwards: 70 From: "Faxes" ;tag=as0980f15f To: ;tag=aetyyy3p04p0F Contact: Call-ID: 6cac8ce45b7ae4644ba7056b742799c7@sipipgw.magrathea.net CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "Faxes" ;privacy=off;screen=yes Content-Length: 0 --- node4*CLI> -- SIP/magrathea-outbound-0a1bd258 answered SIP/87.238.72.155-b7b87118 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> BYE sip:01834861756@92.63.138.97 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 213.166.5.148;branch=z9hG4bK0447.b41174b6.0 Via: SIP/2.0/UDP 213.166.5.139:5070;branch=z9hG4bK0447.188e1726.0 Via: SIP/2.0/UDP 213.166.5.154;rport=5060;branch=z9hG4bK72DrKDD08DNDa Max-Forwards: 68 From: ;tag=aetyyy3p04p0F To: "Faxes" ;tag=as0980f15f Call-ID: 6cac8ce45b7ae4644ba7056b742799c7@sipipgw.magrathea.net CSeq: 121001130 BYE User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Length: 0 <-------------> --- (14 headers 0 lines) --- node4*CLI> Sending to 213.166.5.148 : 5060 (no NAT) <--- Transmitting (no NAT) to 213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 213.166.5.148;branch=z9hG4bK0447.b41174b6.0;received=213.166.5.148 Via: SIP/2.0/UDP 213.166.5.139:5070;branch=z9hG4bK0447.188e1726.0 Via: SIP/2.0/UDP 213.166.5.154;rport=5060;branch=z9hG4bK72DrKDD08DNDa Record-Route: From: ;tag=aetyyy3p04p0F To: "Faxes" ;tag=as0980f15f Call-ID: 6cac8ce45b7ae4644ba7056b742799c7@sipipgw.magrathea.net CSeq: 121001130 BYE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> -- Executing [h@insight-dialout-external:1] Set("SIP/87.238.72.155-b7b87118", "CDR(outbound)=01925413333") in new stack node4*CLI> == Spawn extension (insight-dialout-external, 448447041620, 5) exited non-zero on 'SIP/87.238.72.155-b7b87118' node4*CLI> Scheduling destruction of SIP dialog '752A8F84-AC4511DE-8E6ABA8E-F0DAC5A2@87.238.72.155' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 92.63.138.100, port 5060 Reliably Transmitting (no NAT) to 92.63.138.100:5060: BYE sip:01834861756@87.238.72.155:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK61fc06ce;rport Route: , Max-Forwards: 70 From: ;tag=as7d091253 To: "01834861756" ;tag=A5CC1024-F70 Call-ID: 752A8F84-AC4511DE-8E6ABA8E-F0DAC5A2@87.238.72.155 CSeq: 102 BYE User-Agent: Asterisk PBX 1.6.1.3-rc1 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- node4*CLI> == MixMonitor close filestream node4*CLI> == End MixMonitor Recording SIP/87.238.72.155-b7b87118 node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK61fc06ce;rport=5060 From: ;tag=as7d091253 To: ;tag=A5CC1024-F70 Date: Tue, 29 Sep 2009 15:42:13 gmt Call-ID: 752A8F84-AC4511DE-8E6ABA8E-F0DAC5A2@87.238.72.155 Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 102 BYE <-------------> --- (9 headers 0 lines) --- node4*CLI> SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '6cac8ce45b7ae4644ba7056b742799c7@sipipgw.magrathea.net' Method: BYE Really destroying SIP dialog '752A8F84-AC4511DE-8E6ABA8E-F0DAC5A2@87.238.72.155' Method: ACK node4*CLI> Reliably Transmitting (no NAT) to 213.166.5.148:5060: OPTIONS sip:sipipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK353d241e;rport Max-Forwards: 70 From: "asterisk" ;tag=as7ac2cefd To: Contact: Call-ID: 5424283511e28529656f06f55269dc29@92.63.138.97 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.1.3-rc1 Date: Tue, 29 Sep 2009 15:42:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 403 OPTIONS not supported Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK353d241e;rport=5060 From: "asterisk" ;tag=as7ac2cefd To: ;tag=9a264c9a00f926193bf7ce80aab147c3.49df Call-ID: 5424283511e28529656f06f55269dc29@92.63.138.97 CSeq: 102 OPTIONS Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- node4*CLI> Really destroying SIP dialog '5424283511e28529656f06f55269dc29@92.63.138.97' Method: OPTIONS node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INFO sip:92.63.138.97:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK86ae.40f7f6d3.0 To: sip:92.63.138.97:5060 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-56bd CSeq: 10 INFO Call-ID: 6ad6492a-7670@92.63.138.100 Content-Length: 0 User-Agent: OpenSIPS (1.5.1-notls (i386/linux)) <-------------> --- (8 headers 0 lines) --- node4*CLI> <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK86ae.40f7f6d3.0;received=92.63.138.100 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-56bd To: sip:92.63.138.97:5060;tag=as34641517 Call-ID: 6ad6492a-7670@92.63.138.100 CSeq: 10 INFO Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> BYE sip:448081208743@92.63.138.97:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK4dd7.3e934c64.0 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bK4dd7.60afde4.0 Via: SIP/2.0/UDP 213.166.5.134:5060;branch=z9hG4bK57695321688 From: ;tag=5933CC04-268 To: ;tag=as79859efe Date: Tue, 29 Sep 2009 15:41:12 gmt Call-ID: 59689BD9-AC4511DE-BE80D4B2-5B8D3061@213.166.5.134 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 13 Timestamp: 1254238962 CSeq: 102 BYE Content-Length: 0 <-------------> --- (14 headers 0 lines) --- node4*CLI> Sending to 92.63.138.100 : 5060 (no NAT) <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK4dd7.3e934c64.0;received=92.63.138.100 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bK4dd7.60afde4.0 Via: SIP/2.0/UDP 213.166.5.134:5060;branch=z9hG4bK57695321688 Record-Route: From: ;tag=5933CC04-268 To: ;tag=as79859efe Call-ID: 59689BD9-AC4511DE-BE80D4B2-5B8D3061@213.166.5.134 CSeq: 102 BYE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> -- Executing [h@insight-dialout-external:1] Set("SIP/213.166.5.134-b7b65a58", "CDR(outbound)=02076033170") in new stack node4*CLI> Scheduling destruction of SIP dialog '5b6759c357656c505984f86d3ec90c06@sipipgw.magrathea.net' in 6400 ms (Method: INVITE) node4*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.148, port 5060 node4*CLI> Reliably Transmitting (no NAT) to 213.166.5.148:5060: BYE sip:87.238.72.134 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK206f76dd;rport Route: Max-Forwards: 70 From: "Number-1" ;tag=as69e6e6ee To: ;tag=B69e0v1eS103N Call-ID: 5b6759c357656c505984f86d3ec90c06@sipipgw.magrathea.net CSeq: 103 BYE User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "Number-1" ;privacy=off;screen=yes X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- node4*CLI> == Spawn extension (insight-dialout-external, 448081208743, 5) exited non-zero on 'SIP/213.166.5.134-b7b65a58' node4*CLI> == MixMonitor close filestream == End MixMonitor Recording SIP/213.166.5.134-b7b65a58 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK206f76dd;rport=5060 From: "Number-1" ;tag=as69e6e6ee To: ;tag=B69e0v1eS103N Call-ID: 5b6759c357656c505984f86d3ec90c06@sipipgw.magrathea.net CSeq: 103 BYE User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Length: 0 <-------------> --- (10 headers 0 lines) --- node4*CLI> Really destroying SIP dialog '59689BD9-AC4511DE-BE80D4B2-5B8D3061@213.166.5.134' Method: BYE Really destroying SIP dialog '5b6759c357656c505984f86d3ec90c06@sipipgw.magrathea.net' Method: INVITE node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INFO sip:92.63.138.97:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK46ae.e98d83d2.0 To: sip:92.63.138.97:5060 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-e0eb CSeq: 10 INFO Call-ID: 6ad6492e-7670@92.63.138.100 Content-Length: 0 User-Agent: OpenSIPS (1.5.1-notls (i386/linux)) <-------------> --- (8 headers 0 lines) --- node4*CLI> <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK46ae.e98d83d2.0;received=92.63.138.100 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-e0eb To: sip:92.63.138.97:5060;tag=as0e9718df Call-ID: 6ad6492e-7670@92.63.138.100 CSeq: 10 INFO Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> Reliably Transmitting (no NAT) to 213.166.5.148:5060: OPTIONS sip:sipipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK43301ddd;rport Max-Forwards: 70 From: "asterisk" ;tag=as5e5ccd1a To: Contact: Call-ID: 541631ca5176726421fe50675a7b5def@92.63.138.97 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.1.3-rc1 Date: Tue, 29 Sep 2009 15:43:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 403 OPTIONS not supported Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK43301ddd;rport=5060 From: "asterisk" ;tag=as5e5ccd1a To: ;tag=9a264c9a00f926193bf7ce80aab147c3.dcb7 Call-ID: 541631ca5176726421fe50675a7b5def@92.63.138.97 CSeq: 102 OPTIONS Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- node4*CLI> Really destroying SIP dialog '541631ca5176726421fe50675a7b5def@92.63.138.97' Method: OPTIONS node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> BYE sip:02087154691@92.63.138.97 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 213.166.5.148;branch=z9hG4bKe75e.a7898b05.0 Via: SIP/2.0/UDP 213.166.5.139:5070;branch=z9hG4bKe75e.019cd3d5.0 Via: SIP/2.0/UDP 213.166.5.153;rport=5060;branch=z9hG4bK5NKHU3HpSFmXr Max-Forwards: 68 From: ;tag=D3p6vB8XeQmym To: "Number-2" ;tag=as188c5494 Call-ID: 226c6bf056338ad759023b5c734eeaf7@sipipgw.magrathea.net CSeq: 121001163 BYE User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Length: 0 <-------------> --- (14 headers 0 lines) --- node4*CLI> Sending to 213.166.5.148 : 5060 (no NAT) node4*CLI> <--- Transmitting (no NAT) to 213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 213.166.5.148;branch=z9hG4bKe75e.a7898b05.0;received=213.166.5.148 Via: SIP/2.0/UDP 213.166.5.139:5070;branch=z9hG4bKe75e.019cd3d5.0 Via: SIP/2.0/UDP 213.166.5.153;rport=5060;branch=z9hG4bK5NKHU3HpSFmXr Record-Route: From: ;tag=D3p6vB8XeQmym To: "Number-2" ;tag=as188c5494 Call-ID: 226c6bf056338ad759023b5c734eeaf7@sipipgw.magrathea.net CSeq: 121001163 BYE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> -- Executing [h@insight-dialout-external:1] Set("SIP/213.166.5.133-b7b7d920", "CDR(outbound)=01604878411") in new stack node4*CLI> == Spawn extension (insight-dialout-external, 448450045847, 5) exited non-zero on 'SIP/213.166.5.133-b7b7d920' node4*CLI> Scheduling destruction of SIP dialog 'CED709DD-AC4311DE-8A13EFB9-710EBE4B@213.166.5.133' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 92.63.138.100, port 5060 Reliably Transmitting (no NAT) to 92.63.138.100:5060: BYE sip:02087154691@213.166.5.133:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK66efb5ad;rport Route: , Max-Forwards: 70 From: ;tag=as1743fe35 To: "02087154691" ;tag=65C6DEF8-20C4 Call-ID: CED709DD-AC4311DE-8A13EFB9-710EBE4B@213.166.5.133 CSeq: 102 BYE User-Agent: Asterisk PBX 1.6.1.3-rc1 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- node4*CLI> == MixMonitor close filestream node4*CLI> == End MixMonitor Recording SIP/213.166.5.133-b7b7d920 node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK66efb5ad;rport=5060 From: ;tag=as1743fe35 To: ;tag=65C6DEF8-20C4 Date: Tue, 29 Sep 2009 15:43:20 gmt Call-ID: CED709DD-AC4311DE-8A13EFB9-710EBE4B@213.166.5.133 Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 102 BYE <-------------> --- (9 headers 0 lines) --- node4*CLI> SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog 'CED709DD-AC4311DE-8A13EFB9-710EBE4B@213.166.5.133' Method: ACK Really destroying SIP dialog '226c6bf056338ad759023b5c734eeaf7@sipipgw.magrathea.net' Method: BYE node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INFO sip:92.63.138.97:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKf8be.35209d56.0 To: sip:92.63.138.97:5060 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-98e7 CSeq: 10 INFO Call-ID: 6ad64932-7670@92.63.138.100 Content-Length: 0 User-Agent: OpenSIPS (1.5.1-notls (i386/linux)) <-------------> --- (8 headers 0 lines) --- node4*CLI> <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKf8be.35209d56.0;received=92.63.138.100 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-98e7 To: sip:92.63.138.97:5060;tag=as4d87fb1d Call-ID: 6ad64932-7670@92.63.138.100 CSeq: 10 INFO Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INVITE sip:448005244524@92.63.138.97:5060 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK8c56.93e855d3.0 Via: SIP/2.0/UDP 213.166.5.129;branch=z9hG4bK8c56.d69ca9b5.0 Via: SIP/2.0/UDP 213.166.5.133:5060;branch=z9hG4bK579F3051856 From: "07904407268" ;tag=65D2F9CC-191C To: Date: Tue, 29 Sep 2009 15:43:23 gmt Call-ID: A7AFC61F-AC4511DE-AF25EFB9-710EBE4B@213.166.5.133 Supported: timer,replaces Min-SE: 1800 User-Agent: MSSGW Allow: INVITE, BYE, CANCEL, ACK CSeq: 101 INVITE Max-Forwards: 13 Remote-Party-ID: ;party=calling;screen=yes;privacy=off Timestamp: 1254239003 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 404 v=0 o=CiscoSystemsSIP-GW-UserAgent 2652 3845 IN IP4 213.166.5.133 s=SIP Call c=IN IP4 213.166.5.133 t=0 0 m=audio 17094 RTP/AVP 8 18 4 3 2 0 101 c=IN IP4 213.166.5.133 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:3 GSM/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (23 headers 17 lines) --- node4*CLI> == Using SIP RTP CoS mark 5 node4*CLI> Sending to 92.63.138.100 : 5060 (no NAT) Using INVITE request as basis request - A7AFC61F-AC4511DE-AF25EFB9-710EBE4B@213.166.5.133 No matching peer for '07904407268' from '92.63.138.100:5060' Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 3 Found RTP audio format 2 node4*CLI> Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 213.166.5.133:17094 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Got unsupported a:fmtp in SDP offer Found audio description format G723 for ID 4 Got unsupported a:fmtp in SDP offer Found audio description format GSM for ID 3 Found audio description format G726-32 for ID 2 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8 (alaw), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 213.166.5.133:17094 Looking for 448005244524 in incoming_magrathea (domain 92.63.138.97) node4*CLI> list_route: hop: list_route: hop: node4*CLI> <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK8c56.93e855d3.0;received=92.63.138.100 Via: SIP/2.0/UDP 213.166.5.129;branch=z9hG4bK8c56.d69ca9b5.0 Via: SIP/2.0/UDP 213.166.5.133:5060;branch=z9hG4bK579F3051856 Record-Route: Record-Route: From: "07904407268" ;tag=65D2F9CC-191C To: Call-ID: A7AFC61F-AC4511DE-AF25EFB9-710EBE4B@213.166.5.133 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> node4*CLI> -- Executing AGI("SIP/213.166.5.133-0a18cec0", "agi://web0/track.agi?username=netbasic&campaignName=Protected+Help&campaignId=2361&destination=01329820399") node4*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERPRES()=allowed) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(accountcode)=netbasic) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(userfield)=2361) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERID(name)=Protected Help) node4*CLI> -- AGI Script agi://web0/track.agi?username=netbasic&campaignName=Protected+Help&campaignId=2361&destination=01329820399 completed, returning 0 node4*CLI> -- Executing [448005244524@insight-dialout-external:1] GotoIf("SIP/213.166.5.133-0a18cec0", "0?2:5") in new stack -- Goto (insight-dialout-external,448005244524,5) -- Executing [448005244524@insight-dialout-external:5] Dial("SIP/213.166.5.133-0a18cec0", "SIP/magrathea-outbound/01329820399,40,Cr") in new stack node4*CLI> == Using SIP RTP CoS mark 5 node4*CLI> Audio is at 92.63.138.97 port 17408 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP node4*CLI> Reliably Transmitting (no NAT) to 213.166.5.148:5060: INVITE sip:01329820399@sipipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK6bd2eaa5;rport Max-Forwards: 70 From: "Protected Help" ;tag=as015072e1 To: Contact: Call-ID: 6a85920a353a8c3b36e0ac8162ea13ca@sipipgw.magrathea.net CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "Protected Help" ;privacy=off;screen=yes Date: Tue, 29 Sep 2009 15:43:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 265 v=0 o=root 796275868 796275868 IN IP4 92.63.138.97 s=Asterisk PBX 1.6.1.3-rc1 c=IN IP4 92.63.138.97 t=0 0 m=audio 17408 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called magrathea-outbound/01329820399 <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK8c56.93e855d3.0;received=92.63.138.100 Via: SIP/2.0/UDP 213.166.5.129;branch=z9hG4bK8c56.d69ca9b5.0 Via: SIP/2.0/UDP 213.166.5.133:5060;branch=z9hG4bK579F3051856 Record-Route: Record-Route: From: "07904407268" ;tag=65D2F9CC-191C To: ;tag=as3c9653b1 Call-ID: A7AFC61F-AC4511DE-AF25EFB9-710EBE4B@213.166.5.133 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK6bd2eaa5;rport=5060 From: "Protected Help" ;tag=as015072e1 To: Call-ID: 6a85920a353a8c3b36e0ac8162ea13ca@sipipgw.magrathea.net CSeq: 102 INVITE Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK6bd2eaa5;rport=5060 Record-Route: From: "Protected Help" ;tag=as015072e1 To: ;tag=7DN0KDNSycjrK Call-ID: 6a85920a353a8c3b36e0ac8162ea13ca@sipipgw.magrathea.net CSeq: 102 INVITE Contact: User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Type: application/sdp Content-Length: 180 v=0 o=- 5887946 0 IN IP4 213.166.5.153 s=Cisco SDP 0 c=IN IP4 213.166.5.153 t=0 0 m=audio 47550 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (13 headers 9 lines) --- node4*CLI> Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 213.166.5.153:47550 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 213.166.5.153:47550 -- SIP/magrathea-outbound-0a1bd258 is making progress passing it to SIP/213.166.5.133-0a18cec0 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK6bd2eaa5;rport=5060 Record-Route: From: "Protected Help" ;tag=as015072e1 To: ;tag=7DN0KDNSycjrK Call-ID: 6a85920a353a8c3b36e0ac8162ea13ca@sipipgw.magrathea.net CSeq: 102 INVITE Contact: User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Min-SE: 120 Content-Type: application/sdp Content-Length: 180 v=0 o=- 5887946 0 IN IP4 213.166.5.153 s=Cisco SDP 0 c=IN IP4 213.166.5.153 t=0 0 m=audio 47550 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (14 headers 9 lines) --- node4*CLI> list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.148, port 5060 Transmitting (no NAT) to 213.166.5.148:5060: ACK sip:213.166.5.153 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK1d7fc5ae;rport Route: Max-Forwards: 70 From: "Protected Help" ;tag=as015072e1 To: ;tag=7DN0KDNSycjrK Contact: Call-ID: 6a85920a353a8c3b36e0ac8162ea13ca@sipipgw.magrathea.net CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "Protected Help" ;privacy=off;screen=yes Content-Length: 0 --- node4*CLI> -- SIP/magrathea-outbound-0a1bd258 answered SIP/213.166.5.133-0a18cec0 node4*CLI> Audio is at 92.63.138.97 port 10184 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP node4*CLI> <--- Reliably Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK8c56.93e855d3.0;received=92.63.138.100 Via: SIP/2.0/UDP 213.166.5.129;branch=z9hG4bK8c56.d69ca9b5.0 Via: SIP/2.0/UDP 213.166.5.133:5060;branch=z9hG4bK579F3051856 Record-Route: Record-Route: From: "07904407268" ;tag=65D2F9CC-191C To: ;tag=as3c9653b1 Call-ID: A7AFC61F-AC4511DE-AF25EFB9-710EBE4B@213.166.5.133 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 265 v=0 o=root 458364638 458364638 IN IP4 92.63.138.97 s=Asterisk PBX 1.6.1.3-rc1 c=IN IP4 92.63.138.97 t=0 0 m=audio 10184 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Packet2Packet bridging SIP/213.166.5.133-0a18cec0 and SIP/magrathea-outbound-0a1bd258 node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> ACK sip:448005244524@92.63.138.97:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK8c56.93e855d3.2 Via: SIP/2.0/UDP 213.166.5.129;branch=z9hG4bK8c56.d69ca9b5.2 Via: SIP/2.0/UDP 213.166.5.133:5060;branch=z9hG4bK579F31B1D75 From: ;tag=65D2F9CC-191C To: ;tag=as3c9653b1 Date: Tue, 29 Sep 2009 15:43:23 gmt Call-ID: A7AFC61F-AC4511DE-AF25EFB9-710EBE4B@213.166.5.133 Max-Forwards: 13 CSeq: 101 ACK Content-Length: 0 <-------------> --- (12 headers 0 lines) --- node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INFO sip:92.63.138.97:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKb8be.fedcb5a2.0 To: sip:92.63.138.97:5060 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-2eb1 CSeq: 10 INFO Call-ID: 6ad64936-7670@92.63.138.100 Content-Length: 0 User-Agent: OpenSIPS (1.5.1-notls (i386/linux)) <-------------> --- (8 headers 0 lines) --- node4*CLI> <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKb8be.fedcb5a2.0;received=92.63.138.100 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-2eb1 To: sip:92.63.138.97:5060;tag=as3e3523a8 Call-ID: 6ad64936-7670@92.63.138.100 CSeq: 10 INFO Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> BYE sip:07904407268@92.63.138.97 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 213.166.5.148;branch=z9hG4bKf2b6.1d8e66a.0 Via: SIP/2.0/UDP 213.166.5.139:5070;branch=z9hG4bKf2b6.64133685.0 Via: SIP/2.0/UDP 213.166.5.153;rport=5060;branch=z9hG4bKggS78NS0aFpBp Max-Forwards: 68 From: ;tag=7DN0KDNSycjrK To: "Protected Help" ;tag=as015072e1 Call-ID: 6a85920a353a8c3b36e0ac8162ea13ca@sipipgw.magrathea.net CSeq: 121001182 BYE User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Length: 0 <-------------> --- (14 headers 0 lines) --- node4*CLI> Sending to 213.166.5.148 : 5060 (no NAT) <--- Transmitting (no NAT) to 213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 213.166.5.148;branch=z9hG4bKf2b6.1d8e66a.0;received=213.166.5.148 Via: SIP/2.0/UDP 213.166.5.139:5070;branch=z9hG4bKf2b6.64133685.0 Via: SIP/2.0/UDP 213.166.5.153;rport=5060;branch=z9hG4bKggS78NS0aFpBp Record-Route: From: ;tag=7DN0KDNSycjrK To: "Protected Help" ;tag=as015072e1 Call-ID: 6a85920a353a8c3b36e0ac8162ea13ca@sipipgw.magrathea.net CSeq: 121001182 BYE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> -- Executing [h@insight-dialout-external:1] Set("SIP/213.166.5.133-0a18cec0", "CDR(outbound)=01329820399") in new stack node4*CLI> == Spawn extension (insight-dialout-external, 448005244524, 5) exited non-zero on 'SIP/213.166.5.133-0a18cec0' node4*CLI> Scheduling destruction of SIP dialog 'A7AFC61F-AC4511DE-AF25EFB9-710EBE4B@213.166.5.133' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 92.63.138.100, port 5060 Reliably Transmitting (no NAT) to 92.63.138.100:5060: BYE sip:07904407268@213.166.5.133:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK13ee99ca;rport Route: , Max-Forwards: 70 From: ;tag=as3c9653b1 To: "07904407268" ;tag=65D2F9CC-191C Call-ID: A7AFC61F-AC4511DE-AF25EFB9-710EBE4B@213.166.5.133 CSeq: 102 BYE User-Agent: Asterisk PBX 1.6.1.3-rc1 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK13ee99ca;rport=5060 From: ;tag=as3c9653b1 To: ;tag=65D2F9CC-191C Date: Tue, 29 Sep 2009 15:43:57 gmt Call-ID: A7AFC61F-AC4511DE-AF25EFB9-710EBE4B@213.166.5.133 Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 102 BYE <-------------> --- (9 headers 0 lines) --- node4*CLI> SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog 'A7AFC61F-AC4511DE-AF25EFB9-710EBE4B@213.166.5.133' Method: ACK Really destroying SIP dialog '6a85920a353a8c3b36e0ac8162ea13ca@sipipgw.magrathea.net' Method: BYE node4*CLI> Reliably Transmitting (no NAT) to 213.166.5.148:5060: OPTIONS sip:sipipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK114079b5;rport Max-Forwards: 70 From: "asterisk" ;tag=as754bdf80 To: Contact: Call-ID: 59016efb14a104df00d6f2861b0b687e@92.63.138.97 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.1.3-rc1 Date: Tue, 29 Sep 2009 15:44:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 403 OPTIONS not supported Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK114079b5;rport=5060 From: "asterisk" ;tag=as754bdf80 To: ;tag=9a264c9a00f926193bf7ce80aab147c3.c17e Call-ID: 59016efb14a104df00d6f2861b0b687e@92.63.138.97 CSeq: 102 OPTIONS Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- node4*CLI> Really destroying SIP dialog '59016efb14a104df00d6f2861b0b687e@92.63.138.97' Method: OPTIONS node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INFO sip:92.63.138.97:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK84be.bf0c4bf7.0 To: sip:92.63.138.97:5060 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-90ce CSeq: 10 INFO Call-ID: 6ad6493a-7670@92.63.138.100 Content-Length: 0 User-Agent: OpenSIPS (1.5.1-notls (i386/linux)) <-------------> --- (8 headers 0 lines) --- node4*CLI> <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK84be.bf0c4bf7.0;received=92.63.138.100 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-90ce To: sip:92.63.138.97:5060;tag=as390a4757 Call-ID: 6ad6493a-7670@92.63.138.100 CSeq: 10 INFO Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INVITE sip:448081201315@92.63.138.97:5060 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK4789.ee787927.0 Via: SIP/2.0/UDP 213.166.5.129;branch=z9hG4bK4789.fdad2834.0 Via: SIP/2.0/UDP 213.166.5.133:5060;branch=z9hG4bK579F4D15E2 From: "01707333573" ;tag=65D3E11C-12ED To: Date: Tue, 29 Sep 2009 15:44:22 gmt Call-ID: CAFB850B-AC4511DE-B259EFB9-710EBE4B@213.166.5.133 Supported: timer,replaces Min-SE: 1800 User-Agent: MSSGW Allow: INVITE, BYE, CANCEL, ACK CSeq: 101 INVITE Max-Forwards: 13 Remote-Party-ID: ;party=calling;screen=yes;privacy=off Timestamp: 1254239062 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 403 v=0 o=CiscoSystemsSIP-GW-UserAgent 1781 143 IN IP4 213.166.5.133 s=SIP Call c=IN IP4 213.166.5.133 t=0 0 m=audio 16668 RTP/AVP 8 18 4 3 2 0 101 c=IN IP4 213.166.5.133 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:3 GSM/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (23 headers 17 lines) --- node4*CLI> == Using SIP RTP CoS mark 5 Sending to 92.63.138.100 : 5060 (no NAT) Using INVITE request as basis request - CAFB850B-AC4511DE-B259EFB9-710EBE4B@213.166.5.133 node4*CLI> No matching peer for '01707333573' from '92.63.138.100:5060' Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 3 Found RTP audio format 2 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 213.166.5.133:16668 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Got unsupported a:fmtp in SDP offer Found audio description format G723 for ID 4 Got unsupported a:fmtp in SDP offer F node4*CLI> ound audio description format GSM for ID 3 Found audio description format G726-32 for ID 2 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8 (alaw), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 213.166.5.133:16668 Looking for 448081201315 in incoming_magrathea (domain 92.63.138.97) node4*CLI> list_route: hop: list_route: hop: node4*CLI> <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK4789.ee787927.0;received=92.63.138.100 Via: SIP/2.0/UDP 213.166.5.129;branch=z9hG4bK4789.fdad2834.0 Via: SIP/2.0/UDP 213.166.5.133:5060;branch=z9hG4bK579F4D15E2 Record-Route: Record-Route: From: "01707333573" ;tag=65D3E11C-12ED To: Call-ID: CAFB850B-AC4511DE-B259EFB9-710EBE4B@213.166.5.133 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> node4*CLI> -- Executing AGI("SIP/213.166.5.133-0a18cec0", "agi://web0/track.agi?username=provisioning&campaignName=Number-1&campaignId=379&destination=44800289752&recordCall=yes") node4*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERPRES()=allowed) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(accountcode)=provisioning) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(userfield)=379) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERID(name)=Number-1) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(amaflags)=billing) node4*CLI> -- AGI Script agi://web0/track.agi?username=provisioning&campaignName=Number-1&campaignId=379&destination=44800289752&recordCall=yes completed, returning 0 node4*CLI> -- Executing [448081201315@insight-dialout-external:1] GotoIf("SIP/213.166.5.133-0a18cec0", "1?2:5") in new stack -- Goto (insight-dialout-external,448081201315,2) -- Executing [448081201315@insight-dialout-external:2] Answer("SIP/213.166.5.133-0a18cec0", "") in new stack node4*CLI> Audio is at 92.63.138.97 port 16898 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP node4*CLI> <--- Reliably Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK4789.ee787927.0;received=92.63.138.100 Via: SIP/2.0/UDP 213.166.5.129;branch=z9hG4bK4789.fdad2834.0 Via: SIP/2.0/UDP 213.166.5.133:5060;branch=z9hG4bK579F4D15E2 Record-Route: Record-Route: From: "01707333573" ;tag=65D3E11C-12ED To: ;tag=as3cf2c569 Call-ID: CAFB850B-AC4511DE-B259EFB9-710EBE4B@213.166.5.133 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 267 v=0 o=root 1390761041 1390761041 IN IP4 92.63.138.97 s=Asterisk PBX 1.6.1.3-rc1 c=IN IP4 92.63.138.97 t=0 0 m=audio 16898 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> ACK sip:448081201315@92.63.138.97:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK4789.ee787927.2 Via: SIP/2.0/UDP 213.166.5.129;branch=z9hG4bK4789.fdad2834.2 Via: SIP/2.0/UDP 213.166.5.133:5060;branch=z9hG4bK579F4D2CF0 From: ;tag=65D3E11C-12ED To: ;tag=as3cf2c569 Date: Tue, 29 Sep 2009 15:44:22 gmt Call-ID: CAFB850B-AC4511DE-B259EFB9-710EBE4B@213.166.5.133 Max-Forwards: 13 CSeq: 101 ACK Content-Length: 0 <-------------> --- (12 headers 0 lines) --- node4*CLI> -- Executing [448081201315@insight-dialout-external:3] MixMonitor("SIP/213.166.5.133-0a18cec0", "voip2-1254239062.635.wav") in new stack node4*CLI> -- Executing [448081201315@insight-dialout-external:4] BackGround("SIP/213.166.5.133-0a18cec0", "adinsight-call-recorded") in new stack == Begin MixMonitor Recording SIP/213.166.5.133-0a18cec0 node4*CLI> -- Playing 'adinsight-call-recorded.ulaw' (language 'en') node4*CLI> -- Executing [448081201315@insight-dialout-external:5] Dial("SIP/213.166.5.133-0a18cec0", "SIP/magrathea-outbound/44800289752,40,Cr") in new stack node4*CLI> == Using SIP RTP CoS mark 5 node4*CLI> Audio is at 92.63.138.97 port 16984 node4*CLI> Adding codec 0x8 (alaw) to SDP node4*CLI> Adding non-codec 0x1 (telephone-event) to SDP node4*CLI> Reliably Transmitting (no NAT) to 213.166.5.148:5060: INVITE sip:44800289752@sipipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK1cf6915b;rport Max-Forwards: 70 From: "Number-1" ;tag=as21cb1aeb To: Contact: Call-ID: 3db3c73d482e72f56170316078909b5f@sipipgw.magrathea.net CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "Number-1" ;privacy=off;screen=yes Date: Tue, 29 Sep 2009 15:44:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 265 v=0 o=root 146149831 146149831 IN IP4 92.63.138.97 s=Asterisk PBX 1.6.1.3-rc1 c=IN IP4 92.63.138.97 t=0 0 m=audio 16984 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- node4*CLI> -- Called magrathea-outbound/44800289752 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK1cf6915b;rport=5060 From: "Number-1" ;tag=as21cb1aeb To: Call-ID: 3db3c73d482e72f56170316078909b5f@sipipgw.magrathea.net CSeq: 102 INVITE Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK1cf6915b;rport=5060 From: "Number-1" ;tag=as21cb1aeb To: ;tag=A5CE517C-176E Date: Tue, 29 Sep 2009 15:44:26 gmt Call-ID: 3db3c73d482e72f56170316078909b5f@sipipgw.magrathea.net Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Allow-Events: telephone-event Contact: Record-Route: Content-Disposition: session;handling=required Content-Type: application/sdp Content-Length: 250 v=0 o=CiscoSystemsSIP-GW-UserAgent 2795 6275 IN IP4 87.238.72.155 s=SIP Call c=IN IP4 87.238.72.155 t=0 0 m=audio 18896 RTP/AVP 8 101 c=IN IP4 87.238.72.155 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (15 headers 11 lines) --- node4*CLI> Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 87.238.72.155:18896 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 87.238.72.155:18896 -- SIP/magrathea-outbound-b660f498 is making progress passing it to SIP/213.166.5.133-0a18cec0 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK1cf6915b;rport=5060 From: "Number-1" ;tag=as21cb1aeb To: ;tag=A5CE517C-176E Date: Tue, 29 Sep 2009 15:44:26 gmt Call-ID: 3db3c73d482e72f56170316078909b5f@sipipgw.magrathea.net Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Allow-Events: telephone-event Contact: Record-Route: Content-Type: application/sdp Content-Length: 250 v=0 o=CiscoSystemsSIP-GW-UserAgent 2795 6275 IN IP4 87.238.72.155 s=SIP Call c=IN IP4 87.238.72.155 t=0 0 m=audio 18896 RTP/AVP 8 101 c=IN IP4 87.238.72.155 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (14 headers 11 lines) --- node4*CLI> list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.148, port 5060 Transmitting (no NAT) to 213.166.5.148:5060: ACK sip:0844800289752@87.238.72.155:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK06ed0910;rport Route: Max-Forwards: 70 From: "Number-1" ;tag=as21cb1aeb To: ;tag=A5CE517C-176E Contact: Call-ID: 3db3c73d482e72f56170316078909b5f@sipipgw.magrathea.net CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "Number-1" ;privacy=off;screen=yes Content-Length: 0 --- node4*CLI> -- SIP/magrathea-outbound-b660f498 answered SIP/213.166.5.133-0a18cec0 node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INFO sip:92.63.138.97:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK44be.4174c683.0 To: sip:92.63.138.97:5060 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-2698 CSeq: 10 INFO Call-ID: 6ad6493e-7670@92.63.138.100 Content-Length: 0 User-Agent: OpenSIPS (1.5.1-notls (i386/linux)) <-------------> --- (8 headers 0 lines) --- node4*CLI> <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK44be.4174c683.0;received=92.63.138.100 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-2698 To: sip:92.63.138.97:5060;tag=as1ea690f3 Call-ID: 6ad6493e-7670@92.63.138.100 CSeq: 10 INFO Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> -- Remote UNIX connection disconnected node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INVITE sip:442033933660@92.63.138.97:5060 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK9ff3.fc3858c4.0 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bK9ff3.aa142674.0 Via: SIP/2.0/UDP 213.166.5.140:5060;branch=z9hG4bK50417D61A Remote-Party-ID: ;party=calling;screen=yes;privacy=off From: "07837689027" ;tag=C66BE99C-269C To: Date: Tue, 29 Sep 2009 15:45:05 gmt Call-ID: E484B7F4-AC4511DE-9AA2D31F-EFD16753@213.166.5.140 Supported: timer,resource-priority,replaces Min-SE: 1800 User-Agent: MSSGW Allow: INVITE, BYE, CANCEL, ACK CSeq: 101 INVITE Max-Forwards: 13 Timestamp: 1254239105 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 418 v=0 o=CiscoSystemsSIP-GW-UserAgent 5520 3034 IN IP4 213.166.5.140 s=SIP Call c=IN IP4 213.166.5.140 t=0 0 m=audio 19748 RTP/AVP 8 18 4 3 98 0 101 c=IN IP4 213.166.5.140 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:4 G723/8000 a=fmtp:4 bitrate=6.3;annexa=no a=rtpmap:3 GSM/8000 a=rtpmap:98 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (24 headers 17 lines) --- node4*CLI> == Using SIP RTP CoS mark 5 Sending to 92.63.138.100 : 5060 (no NAT) Using INVITE request as basis request - E484B7F4-AC4511DE-9AA2D31F-EFD16753@213.166.5.140 No matching peer for '07837689027' from '92.63.138.100:5060' Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 3 Found RTP audio format 98 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 213.166.5.140:19748 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Got unsupported a:fmtp in SDP offer Found audio description format G723 for ID 4 Got unsupported a:fmtp in SDP offer Found audio description format GSM for ID 3 Found audio description format G726-32 for ID 98 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8 (alaw), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 213.166.5.140:19748 Looking for 442033933660 in incoming_magrathea (domain 92.63.138.97) node4*CLI> list_route: hop: list_route: hop: node4*CLI> <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK9ff3.fc3858c4.0;received=92.63.138.100 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bK9ff3.aa142674.0 Via: SIP/2.0/UDP 213.166.5.140:5060;branch=z9hG4bK50417D61A Record-Route: Record-Route: From: "07837689027" ;tag=C66BE99C-269C To: C node4*CLI> all-ID: E484B7F4-AC4511DE-9AA2D31F-EFD16753@213.166.5.140 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> node4*CLI> -- Executing AGI("SIP/213.166.5.140-b662a6e0", "agi://web0/track.agi?username=provisioning&campaignName=Number-1&campaignId=381&destination=02085008811&recordCall=yes") node4*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERPRES()=allowed) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(accountcode)=provisioning) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(userfield)=381) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERID(name)=Number-1) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(amaflags)=billing) node4*CLI> -- AGI Script agi://web0/track.agi?username=provisioning&campaignName=Number-1&campaignId=381&destination=02085008811&recordCall=yes completed, returning 0 node4*CLI> -- Executing [442033933660@insight-dialout-external:1] GotoIf("SIP/213.166.5.140-b662a6e0", "1?2:5") in new stack node4*CLI> -- Goto (insight-dialout-external,442033933660,2) -- Executing [442033933660@insight-dialout-external:2] Answer("SIP/213.166.5.140-b662a6e0", "") in new stack node4*CLI> Audio is at 92.63.138.97 port 13116 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK9ff3.fc3858c4.0;received=92.63.138.100 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bK9ff3.aa142674.0 Via: SIP/2.0/UDP 213.166.5.140:5060;branch=z9hG4bK50417D61A Record-Route: Record-Route: From: "07837689027" ;tag=C66BE99C-269C To: ;tag=as77f20c09 Call-ID: E484B7F4-AC4511DE-9AA2D31F-EFD16753@213.166.5.140 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 265 v=0 o=root 283371028 283371028 IN IP4 92.63.138.97 s=Asterisk PBX 1.6.1.3-rc1 c=IN IP4 92.63.138.97 t=0 0 m=audio 13116 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> ACK sip:442033933660@92.63.138.97:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK9ff3.fc3858c4.2 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bK9ff3.aa142674.2 Via: SIP/2.0/UDP 213.166.5.140:5060;branch=z9hG4bK50417EA2E From: ;tag=C66BE99C-269C To: ;tag=as77f20c09 Date: Tue, 29 Sep 2009 15:45:05 gmt Call-ID: E484B7F4-AC4511DE-9AA2D31F-EFD16753@213.166.5.140 Max-Forwards: 13 CSeq: 101 ACK Allow-Events: telephone-event Content-Length: 0 <-------------> --- (13 headers 0 lines) --- node4*CLI> -- Executing [442033933660@insight-dialout-external:3] MixMonitor("SIP/213.166.5.140-b662a6e0", "voip2-1254239105.637.wav") in new stack node4*CLI> == Begin MixMonitor Recording SIP/213.166.5.140-b662a6e0 node4*CLI> -- Executing [442033933660@insight-dialout-external:4] BackGround("SIP/213.166.5.140-b662a6e0", "adinsight-call-recorded") in new stack node4*CLI> -- Playing 'adinsight-call-recorded.ulaw' (language 'en') node4*CLI> -- Executing [442033933660@insight-dialout-external:5] Dial("SIP/213.166.5.140-b662a6e0", "SIP/magrathea-outbound/02085008811,40,Cr") in new stack node4*CLI> == Using SIP RTP CoS mark 5 node4*CLI> Audio is at 92.63.138.97 port 13248 node4*CLI> Adding codec 0x8 (alaw) to SDP node4*CLI> Adding non-codec 0x1 (telephone-event) to SDP node4*CLI> Reliably Transmitting (no NAT) to 213.166.5.148:5060: INVITE sip:02085008811@sipipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK24c56cd7;rport Max-Forwards: 70 From: "Number-1" ;tag=as77a3588f To: Contact: Call-ID: 503c745d40bb85fe698580104885843a@sipipgw.magrathea.net CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "Number-1" ;privacy=off;screen=yes Date: Tue, 29 Sep 2009 15:45:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 267 v=0 o=root 1419592261 1419592261 IN IP4 92.63.138.97 s=Asterisk PBX 1.6.1.3-rc1 c=IN IP4 92.63.138.97 t=0 0 m=audio 13248 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- node4*CLI> -- Called magrathea-outbound/02085008811 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK24c56cd7;rport=5060 From: "Number-1" ;tag=as77a3588f To: Call-ID: 503c745d40bb85fe698580104885843a@sipipgw.magrathea.net CSeq: 102 INVITE Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK24c56cd7;rport=5060 Record-Route: From: "Number-1" ;tag=as77a3588f To: ;tag=HmXX9KN9evK5e Call-ID: 503c745d40bb85fe698580104885843a@sipipgw.magrathea.net CSeq: 102 INVITE Contact: User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Type: application/sdp Content-Length: 181 v=0 o=- 53373078 0 IN IP4 87.238.72.134 s=Cisco SDP 0 c=IN IP4 87.238.72.134 t=0 0 m=audio 36642 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (13 headers 9 lines) --- node4*CLI> Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 87.238.72.134:36642 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 87.238.72.134:36642 node4*CLI> -- SIP/magrathea-outbound-0a180578 is making progress passing it to SIP/213.166.5.140-b662a6e0 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK24c56cd7;rport=5060 Record-Route: From: "Number-1" ;tag=as77a3588f To: ;tag=HmXX9KN9evK5e Call-ID: 503c745d40bb85fe698580104885843a@sipipgw.magrathea.net CSeq: 102 INVITE Contact: User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Min-SE: 120 Content-Type: application/sdp Content-Length: 181 v=0 o=- 53373078 0 IN IP4 87.238.72.134 s=Cisco SDP 0 c=IN IP4 87.238.72.134 t=0 0 m=audio 36642 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (14 headers 9 lines) --- node4*CLI> list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.148, port 5060 Transmitting (no NAT) to 213.166.5.148:5060: ACK sip:87.238.72.134 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK6acec734;rport Route: Max-Forwards: 70 From: "Number-1" ;tag=as77a3588f To: ;tag=HmXX9KN9evK5e Contact: Call-ID: 503c745d40bb85fe698580104885843a@sipipgw.magrathea.net CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "Number-1" ;privacy=off;screen=yes Content-Length: 0 --- node4*CLI> -- SIP/magrathea-outbound-0a180578 answered SIP/213.166.5.140-b662a6e0 node4*CLI> Reliably Transmitting (no NAT) to 213.166.5.148:5060: OPTIONS sip:sipipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK5c0e6667;rport Max-Forwards: 70 From: "asterisk" ;tag=as1f59f20f To: Contact: Call-ID: 5ac8c25c15cff57f3bb1aa6b30c3fad1@92.63.138.97 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.1.3-rc1 Date: Tue, 29 Sep 2009 15:45:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 403 OPTIONS not supported Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK5c0e6667;rport=5060 From: "asterisk" ;tag=as1f59f20f To: ;tag=9a264c9a00f926193bf7ce80aab147c3.5d63 Call-ID: 5ac8c25c15cff57f3bb1aa6b30c3fad1@92.63.138.97 CSeq: 102 OPTIONS Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- node4*CLI> Really destroying SIP dialog '5ac8c25c15cff57f3bb1aa6b30c3fad1@92.63.138.97' Method: OPTIONS node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INFO sip:92.63.138.97:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKfe8e.6f2c3411.0 To: sip:92.63.138.97:5060 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-d8af CSeq: 10 INFO Call-ID: 6ad64942-7670@92.63.138.100 Content-Length: 0 User-Agent: OpenSIPS (1.5.1-notls (i386/linux)) <-------------> --- (8 headers 0 lines) --- node4*CLI> <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKfe8e.6f2c3411.0;received=92.63.138.100 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-d8af To: sip:92.63.138.97:5060;tag=as206f2936 Call-ID: 6ad64942-7670@92.63.138.100 CSeq: 10 INFO Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INVITE sip:442034118212@92.63.138.97:5060 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKa1f.7a406b56.0 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bKa1f.4a50bcb4.0 Via: SIP/2.0/UDP 213.166.5.140:5060;branch=z9hG4bK5041D4195 From: "anonymous" ;tag=C66C5284-208F To: Date: Tue, 29 Sep 2009 15:45:32 gmt Call-ID: F486A1D5-AC4511DE-9BD6D31F-EFD16753@213.166.5.140 Supported: timer,resource-priority,replaces Min-SE: 1800 User-Agent: MSSGW Allow: INVITE, BYE, CANCEL, ACK CSeq: 101 INVITE Max-Forwards: 13 Timestamp: 1254239132 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 417 v=0 o=CiscoSystemsSIP-GW-UserAgent 8500 421 IN IP4 213.166.5.140 s=SIP Call c=IN IP4 213.166.5.140 t=0 0 m=audio 16490 RTP/AVP 8 18 4 3 98 0 101 c=IN IP4 213.166.5.140 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:4 G723/8000 a=fmtp:4 bitrate=6.3;annexa=no a=rtpmap:3 GSM/8000 a=rtpmap:98 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> node4*CLI> --- (23 headers 17 lines) --- node4*CLI> == Using SIP RTP CoS mark 5 node4*CLI> Sending to 92.63.138.100 : 5060 (no NAT) Using INVITE request as basis request - F486A1D5-AC4511DE-9BD6D31F-EFD16753@213.166.5.140 No matching peer for '213.166.5.140' from '92.63.138.100:5060' Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 3 Found RTP audio format 98 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 213.166.5.140:16490 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Got unsupported a:fmtp in SDP offer Found audio description format G723 for ID 4 Got unsupported a:fmtp in SDP offer Found audio description format GSM for ID 3 Found audio description format G726-32 for ID 98 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8 (alaw), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 213.166.5.140:16490 Looking for 442034118212 in incoming_magrathea (domain 92.63.138.97) node4*CLI> list_route: hop: list_route: hop: node4*CLI> <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKa1f.7a406b56.0;received=92.63.138.100 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bKa1f.4a50bcb4.0 Via: SIP/2.0/UDP 213.166.5.140:5060;branch=z9hG4bK5041D4195 Record-Route: Record-Route: From: "anonymous" ;tag=C66C5284-208F To: Call-ID: F486A1D5-AC4511DE-9BD6D31F-EFD16753@213.166.5.140 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> node4*CLI> -- Executing AGI("SIP/213.166.5.140-b662fa10", "agi://web0/track.agi?username=provisioning&campaignName=Number-1&campaignId=825&destination=01708443551&recordCall=yes") node4*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERPRES()=allowed) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(accountcode)=provisioning) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(userfield)=825) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERID(name)=Number-1) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(amaflags)=billing) node4*CLI> -- AGI Script agi://web0/track.agi?username=provisioning&campaignName=Number-1&campaignId=825&destination=01708443551&recordCall=yes completed, returning 0 node4*CLI> -- Executing [442034118212@insight-dialout-external:1] GotoIf("SIP/213.166.5.140-b662fa10", "1?2:5") in new stack node4*CLI> -- Goto (insight-dialout-external,442034118212,2) -- Executing [442034118212@insight-dialout-external:2] Answer("SIP/213.166.5.140-b662fa10", "") in new stack node4*CLI> Audio is at 92.63.138.97 port 14176 Adding codec 0x8 (alaw) to SDP node4*CLI> Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKa1f.7a406b56.0;received=92.63.138.100 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bKa1f.4a50bcb4.0 Via: SIP/2.0/UDP 213.166.5.140:5060;branch=z9hG4bK5041D4195 Record-Route: Record-Route: From: "anonymous" ;tag=C66C5284-208F To: ;tag=as253f7d2b Call-ID: F486A1D5-AC4511DE-9BD6D31F-EFD16753@213.166.5.140 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 267 v=0 o=root 1680072518 1680072518 IN IP4 92.63.138.97 s=Asterisk PBX 1.6.1.3-rc1 c=IN IP4 92.63.138.97 t=0 0 m=audio 14176 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> ACK sip:442034118212@92.63.138.97:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKa1f.7a406b56.2 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bKa1f.4a50bcb4.2 Via: SIP/2.0/UDP 213.166.5.140:5060;branch=z9hG4bK5041D5CAB From: "anonymous" ;tag=C66C5284-208F To: ;tag=as253f7d2b Date: Tue, 29 Sep 2009 15:45:32 gmt Call-ID: F486A1D5-AC4511DE-9BD6D31F-EFD16753@213.166.5.140 Max-Forwards: 13 CSeq: 101 ACK Allow-Events: telephone-event Content-Length: 0 <-------------> --- (13 headers 0 lines) --- node4*CLI> -- Executing [442034118212@insight-dialout-external:3] MixMonitor("SIP/213.166.5.140-b662fa10", "voip2-1254239132.639.wav") in new stack node4*CLI> -- Executing [442034118212@insight-dialout-external:4] BackGround("SIP/213.166.5.140-b662fa10", "adinsight-call-recorded") in new stack node4*CLI> == Begin MixMonitor Recording SIP/213.166.5.140-b662fa10 node4*CLI> -- Playing 'adinsight-call-recorded.ulaw' (language 'en') node4*CLI> -- Executing [442034118212@insight-dialout-external:5] Dial("SIP/213.166.5.140-b662fa10", "SIP/magrathea-outbound/01708443551,40,Cr") in new stack node4*CLI> == Using SIP RTP CoS mark 5 node4*CLI> Audio is at 92.63.138.97 port 11632 node4*CLI> Adding codec 0x8 (alaw) to SDP node4*CLI> Adding non-codec 0x1 (telephone-event) to SDP node4*CLI> Reliably Transmitting (no NAT) to 213.166.5.148:5060: INVITE sip:01708443551@sipipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK4ce75f3e;rport Max-Forwards: 70 From: "Number-1" ;tag=as17d751b6 To: Contact: Call-ID: 55c5cf6d6d89fec63b61df7e63ee0163@sipipgw.magrathea.net CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "Number-1" ;privacy=off;screen=yes Date: Tue, 29 Sep 2009 15:45:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 265 v=0 o=root 662827072 662827072 IN IP4 92.63.138.97 s=Asterisk PBX 1.6.1.3-rc1 c=IN IP4 92.63.138.97 t=0 0 m=audio 11632 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- node4*CLI> -- Called magrathea-outbound/01708443551 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK4ce75f3e;rport=5060 From: "Number-1" ;tag=as17d751b6 To: Call-ID: 55c5cf6d6d89fec63b61df7e63ee0163@sipipgw.magrathea.net CSeq: 102 INVITE Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK4ce75f3e;rport=5060 Record-Route: From: "Number-1" ;tag=as17d751b6 To: ;tag=9r8ZXje8KFNHN Call-ID: 55c5cf6d6d89fec63b61df7e63ee0163@sipipgw.magrathea.net CSeq: 102 INVITE Contact: User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Length: 0 <-------------> --- (12 headers 0 lines) --- node4*CLI> -- SIP/magrathea-outbound-b6634758 is ringing node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK4ce75f3e;rport=5060 Record-Route: From: "Number-1" ;tag=as17d751b6 To: ;tag=9r8ZXje8KFNHN Call-ID: 55c5cf6d6d89fec63b61df7e63ee0163@sipipgw.magrathea.net CSeq: 102 INVITE Contact: User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Length: 0 <-------------> --- (12 headers 0 lines) --- node4*CLI> -- SIP/magrathea-outbound-b6634758 is ringing node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK4ce75f3e;rport=5060 Record-Route: From: "Number-1" ;tag=as17d751b6 To: ;tag=9r8ZXje8KFNHN Call-ID: 55c5cf6d6d89fec63b61df7e63ee0163@sipipgw.magrathea.net CSeq: 102 INVITE Contact: User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Min-SE: 120 Content-Type: application/sdp Content-Length: 233 v=0 o=root 3405 3405 IN IP4 87.238.72.134 s=session c=IN IP4 87.238.72.134 t=0 0 m=audio 37178 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=direction:active <-------------> --- (14 headers 12 lines) --- node4*CLI> Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 87.238.72.134:37178 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 87.238.72.134:37178 node4*CLI> list_route: hop: node4*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.148, port 5060 Transmitting (no NAT) to 213.166.5.148:5060: ACK sip:87.238.72.134 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK6a62f6a1;rport Route: Max-Forwards: 70 From: "Number-1" ;tag=as17d751b6 To: ;tag=9r8ZXje8KFNHN Contact: Call-ID: 55c5cf6d6d89fec63b61df7e63ee0163@sipipgw.magrathea.net CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "Number-1" ;privacy=off;screen=yes Content-Length: 0 --- node4*CLI> -- SIP/magrathea-outbound-b6634758 answered SIP/213.166.5.140-b662fa10 node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INFO sip:92.63.138.97:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKbe8e.9e4b2692.0 To: sip:92.63.138.97:5060 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-6ef9 CSeq: 10 INFO Call-ID: 6ad64946-7670@92.63.138.100 Content-Length: 0 User-Agent: OpenSIPS (1.5.1-notls (i386/linux)) <-------------> --- (8 headers 0 lines) --- node4*CLI> <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKbe8e.9e4b2692.0;received=92.63.138.100 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-6ef9 To: sip:92.63.138.97:5060;tag=as42f95470 Call-ID: 6ad64946-7670@92.63.138.100 CSeq: 10 INFO Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INVITE sip:448447747777@92.63.138.97:5060 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK2c3d.bbfa7d96.0 Via: SIP/2.0/UDP 87.238.72.153;branch=z9hG4bK2c3d.69082027.0 Via: SIP/2.0/UDP 87.238.72.149:5060;branch=z9hG4bK1694C69AB6 From: "01455554861" ;tag=B0D61520-1BBC To: Date: Tue, 29 Sep 2009 15:46:17 gmt Call-ID: F4E5DAE-AC4611DE-BB83F3EC-65114225@87.238.72.149 Supported: timer,replaces Min-SE: 1800 User-Agent: MSSGW(B) Allow: INVITE, BYE, CANCEL, ACK CSeq: 101 INVITE Max-Forwards: 13 Remote-Party-ID: ;party=calling;screen=yes;privacy=off Timestamp: 1254239177 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 403 v=0 o=CiscoSystemsSIP-GW-UserAgent 4880 418 IN IP4 87.238.72.149 s=SIP Call c=IN IP4 87.238.72.149 t=0 0 m=audio 19108 RTP/AVP 8 18 4 3 2 0 101 c=IN IP4 87.238.72.149 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:3 GSM/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (23 headers 17 lines) --- node4*CLI> == Using SIP RTP CoS mark 5 Sending to 92.63.138.100 : 5060 (no NAT) Using INVITE request as basis request - F4E5DAE-AC4611DE-BB83F3EC-65114225@87.238.72.149 node4*CLI> No matching peer for '01455554861' from '92.63.138.100:5060' node4*CLI> Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 3 node4*CLI> Found RTP audio format 2 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 87.238.72.149:19108 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Got unsupported a:fmtp in SDP offer Found audio description format G723 for ID 4 Got unsupported a:fmtp in SDP offer Found audio description format GSM for ID 3 Found audio description format G726-32 for ID 2 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8 (alaw), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) node4*CLI> Peer audio RTP is at port 87.238.72.149:19108 Looking for 448447747777 in incoming_magrathea (domain 92.63.138.97) node4*CLI> list_route: hop: list_route: hop: node4*CLI> <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK2c3d.bbfa7d96.0;received=92.63.138.100 Via: SIP/2.0/UDP 87.238.72.153;branch=z9hG4bK2c3d.69082027.0 Via: SIP/2.0/UDP 87.238.72.149:5060;branch=z9hG4bK1694C69AB6 Record-Route: Record-Route: From: "01455554861" ;tag=B0D61520-1BBC To: Call-ID: F4E5DAE-AC4611DE-BB83F3EC-65114225@87.238.72.149 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> node4*CLI> -- Executing AGI("SIP/87.238.72.149-b7b87118", "agi://web0/track.agi?username=freestart office&campaignName=freestart+main+number&campaignId=841&destination=01942406100&recordCall=yes") node4*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERPRES()=allowed) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(accountcode)=freestart office) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(userfield)=841) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERID(name)=freestart main number) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(amaflags)=billing) node4*CLI> -- AGI Script agi://web0/track.agi?username=freestart office&campaignName=freestart+main+number&campaignId=841&destination=01942406100&recordCall=yes completed, returning 0 node4*CLI> -- Executing [448447747777@insight-dialout-external:1] GotoIf("SIP/87.238.72.149-b7b87118", "1?2:5") in new stack node4*CLI> -- Goto (insight-dialout-external,448447747777,2) -- Executing [448447747777@insight-dialout-external:2] Answer("SIP/87.238.72.149-b7b87118", "") in new stack node4*CLI> Audio is at 92.63.138.97 port 13026 node4*CLI> Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK2c3d.bbfa7d96.0;received=92.63.138.100 Via: SIP/2.0/UDP 87.238.72.153;branch=z9hG4bK2c3d.69082027.0 Via: SIP/2.0/UDP 87.238.72.149:5060;branch=z9hG4bK1694C69AB6 Record-Route: Record-Route: From: "01455554861" ;tag=B0D61520-1BBC To: ;tag=as2661c4ae Call-ID: F4E5DAE-AC4611DE-BB83F3EC-65114225@87.238.72.149 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 267 v=0 o=root 1636459550 1636459550 IN IP4 92.63.138.97 s=Asterisk PBX 1.6.1.3-rc1 c=IN IP4 92.63.138.97 t=0 0 m=audio 13026 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> ACK sip:448447747777@92.63.138.97:5060 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK2c3d.bbfa7d96.2 Via: SIP/2.0/UDP 87.238.72.153;branch=z9hG4bK2c3d.69082027.2 Via: SIP/2.0/UDP 87.238.72.149:5060;branch=z9hG4bK1694C6B1B9 From: ;tag=B0D61520-1BBC To: ;tag=as2661c4ae Date: Tue, 29 Sep 2009 15:46:17 gmt Call-ID: F4E5DAE-AC4611DE-BB83F3EC-65114225@87.238.72.149 Max-Forwards: 13 CSeq: 101 ACK Content-Length: 0 <-------------> --- (13 headers 0 lines) --- node4*CLI> -- Executing [448447747777@insight-dialout-external:3] MixMonitor("SIP/87.238.72.149-b7b87118", "voip2-1254239177.641.wav") in new stack node4*CLI> -- Executing [448447747777@insight-dialout-external:4] BackGround("SIP/87.238.72.149-b7b87118", "adinsight-call-recorded") in new stack node4*CLI> == Begin MixMonitor Recording SIP/87.238.72.149-b7b87118 node4*CLI> -- Playing 'adinsight-call-recorded.ulaw' (language 'en') node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> BYE sip:448005244845@92.63.138.97:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK6b47.a77906d.0 Via: SIP/2.0/UDP 213.166.5.129;branch=z9hG4bK6b47.e7094502.0 Via: SIP/2.0/UDP 213.166.5.133:5060;branch=z9hG4bK579F811110D From: ;tag=65CE9D54-16D4 To: ;tag=as198a2824 Date: Tue, 29 Sep 2009 15:38:37 gmt Call-ID: FD53D4DF-AC4411DE-A123EFB9-710EBE4B@213.166.5.133 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 13 Timestamp: 1254239178 CSeq: 102 BYE Reason: Q.850;cause=16 Content-Length: 0 <-------------> --- (15 headers 0 lines) --- Sending to 92.63.138.100 : 5060 (no NAT) node4*CLI> <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK6b47.a77906d.0;received=92.63.138.100 Via: SIP/2.0/UDP 213.166.5.129;branch=z9hG4bK6b47.e7094502.0 Via: SIP/2.0/UDP 213.166.5.133:5060;branch=z9hG4bK579F811110D Record-Route: From: ;tag=65CE9D54-16D4 To: ;tag=as198a2824 Call-ID: FD53D4DF-AC4411DE-A123EFB9-710EBE4B@213.166.5.133 CSeq: 102 BYE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> -- Executing [h@insight-dialout-external:1] Set("SIP/213.166.5.133-b7b56408", "CDR(outbound)=08450744027") in new stack node4*CLI> Scheduling destruction of SIP dialog '7c005fd16999b588292241f92a2d17a5@sipipgw.magrathea.net' in 6400 ms (Method: INVITE) node4*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.148, port 5060 node4*CLI> Reliably Transmitting (no NAT) to 213.166.5.148:5060: BYE sip:08448450744027@87.238.72.155:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK5f933ded;rport Route: Max-Forwards: 70 From: "Affiliate 1" ;tag=as2ed7579c To: ;tag=A5C8FEF8-1122 Call-ID: 7c005fd16999b588292241f92a2d17a5@sipipgw.magrathea.net CSeq: 103 BYE User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "Affiliate 1" ;privacy=off;screen=yes X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- node4*CLI> == Spawn extension (insight-dialout-external, 448005244845, 5) exited non-zero on 'SIP/213.166.5.133-b7b56408' node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK5f933ded;rport=5060 From: "Affiliate 1" ;tag=as2ed7579c To: ;tag=A5C8FEF8-1122 Date: Tue, 29 Sep 2009 15:46:18 gmt Call-ID: 7c005fd16999b588292241f92a2d17a5@sipipgw.magrathea.net Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 103 BYE <-------------> --- (9 headers 0 lines) --- node4*CLI> Really destroying SIP dialog '7c005fd16999b588292241f92a2d17a5@sipipgw.magrathea.net' Method: INVITE node4*CLI> Really destroying SIP dialog 'FD53D4DF-AC4411DE-A123EFB9-710EBE4B@213.166.5.133' Method: BYE node4*CLI> Reliably Transmitting (no NAT) to 213.166.5.148:5060: OPTIONS sip:sipipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK2d6aac74;rport Max-Forwards: 70 From: "asterisk" ;tag=as2eca4bea To: Contact: Call-ID: 7092393b5b0caff4130e4d0a64797917@92.63.138.97 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.1.3-rc1 Date: Tue, 29 Sep 2009 15:46:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 403 OPTIONS not supported Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK2d6aac74;rport=5060 From: "asterisk" ;tag=as2eca4bea To: ;tag=9a264c9a00f926193bf7ce80aab147c3.30b4 Call-ID: 7092393b5b0caff4130e4d0a64797917@92.63.138.97 CSeq: 102 OPTIONS Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- node4*CLI> Really destroying SIP dialog '7092393b5b0caff4130e4d0a64797917@92.63.138.97' Method: OPTIONS node4*CLI> -- Executing [448447747777@insight-dialout-external:5] Dial("SIP/87.238.72.149-b7b87118", "SIP/magrathea-outbound/01942406100,40,Cr") in new stack node4*CLI> == Using SIP RTP CoS mark 5 node4*CLI> Audio is at 92.63.138.97 port 10642 node4*CLI> Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP node4*CLI> Reliably Transmitting (no NAT) to 213.166.5.148:5060: INVITE sip:01942406100@sipipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK3e760003;rport Max-Forwards: 70 From: "freestart main number" ;tag=as0c6dbab3 To: Contact: Call-ID: 3c8996672e1cc9f330e9a6cb27130d09@sipipgw.magrathea.net CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "freestart main number" ;privacy=off;screen=yes Date: Tue, 29 Sep 2009 15:46:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 267 v=0 o=root 1616482609 1616482609 IN IP4 92.63.138.97 s=Asterisk PBX 1.6.1.3-rc1 c=IN IP4 92.63.138.97 t=0 0 m=audio 10642 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- node4*CLI> -- Called magrathea-outbound/01942406100 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK3e760003;rport=5060 From: "freestart main number" ;tag=as0c6dbab3 To: Call-ID: 3c8996672e1cc9f330e9a6cb27130d09@sipipgw.magrathea.net CSeq: 102 INVITE Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INFO sip:92.63.138.97:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK829e.b075ed53.0 To: sip:92.63.138.97:5060 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-d086 CSeq: 10 INFO Call-ID: 6ad6494a-7670@92.63.138.100 Content-Length: 0 User-Agent: OpenSIPS (1.5.1-notls (i386/linux)) <-------------> --- (8 headers 0 lines) --- node4*CLI> <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK829e.b075ed53.0;received=92.63.138.100 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-d086 To: sip:92.63.138.97:5060;tag=as63d2270e Call-ID: 6ad6494a-7670@92.63.138.100 CSeq: 10 INFO Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK3e760003;rport=5060 Record-Route: From: "freestart main number" ;tag=as0c6dbab3 To: ;tag=gtevK2ap5N5BH Call-ID: 3c8996672e1cc9f330e9a6cb27130d09@sipipgw.magrathea.net CSeq: 102 INVITE Contact: User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Type: application/sdp Content-Length: 181 v=0 o=- 38572451 0 IN IP4 213.166.5.153 s=Cisco SDP 0 c=IN IP4 213.166.5.153 t=0 0 m=audio 31438 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (13 headers 9 lines) --- node4*CLI> Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 213.166.5.153:31438 node4*CLI> Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer node4*CLI> Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 213.166.5.153:31438 node4*CLI> -- SIP/magrathea-outbound-0a19dd40 is making progress passing it to SIP/87.238.72.149-b7b87118 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK3e760003;rport=5060 Record-Route: From: "freestart main number" ;tag=as0c6dbab3 To: ;tag=gtevK2ap5N5BH Call-ID: 3c8996672e1cc9f330e9a6cb27130d09@sipipgw.magrathea.net CSeq: 102 INVITE Contact: User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Min-SE: 120 Content-Type: application/sdp Content-Length: 181 v=0 o=- 38572451 0 IN IP4 213.166.5.153 s=Cisco SDP 0 c=IN IP4 213.166.5.153 t=0 0 m=audio 31438 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (14 headers 9 lines) --- node4*CLI> list_route: hop: node4*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.148, port 5060 Transmitting (no NAT) to 213.166.5.148:5060: ACK sip:213.166.5.153 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK04b80355;rport Route: Max-Forwards: 70 From: "freestart main number" ;tag=as0c6dbab3 To: ;tag=gtevK2ap5N5BH Contact: Call-ID: 3c8996672e1cc9f330e9a6cb27130d09@sipipgw.magrathea.net CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "freestart main number" ;privacy=off;screen=yes Content-Length: 0 --- node4*CLI> -- SIP/magrathea-outbound-0a19dd40 answered SIP/87.238.72.149-b7b87118 node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INFO sip:92.63.138.97:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK429e.dae74f2.0 To: sip:92.63.138.97:5060 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-66d0 CSeq: 10 INFO Call-ID: 6ad6494e-7670@92.63.138.100 Content-Length: 0 User-Agent: OpenSIPS (1.5.1-notls (i386/linux)) <-------------> --- (8 headers 0 lines) --- node4*CLI> <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK429e.dae74f2.0;received=92.63.138.100 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-66d0 To: sip:92.63.138.97:5060;tag=as1ba1d93d Call-ID: 6ad6494e-7670@92.63.138.100 CSeq: 10 INFO Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> Reliably Transmitting (no NAT) to 213.166.5.148:5060: OPTIONS sip:sipipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK06e39111;rport Max-Forwards: 70 From: "asterisk" ;tag=as7191e630 To: Contact: Call-ID: 776d32be46bfe97f5a8fe578665b9079@92.63.138.97 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.1.3-rc1 Date: Tue, 29 Sep 2009 15:47:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 403 OPTIONS not supported Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK06e39111;rport=5060 From: "asterisk" ;tag=as7191e630 To: ;tag=9a264c9a00f926193bf7ce80aab147c3.60f2 Call-ID: 776d32be46bfe97f5a8fe578665b9079@92.63.138.97 CSeq: 102 OPTIONS Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- node4*CLI> Really destroying SIP dialog '776d32be46bfe97f5a8fe578665b9079@92.63.138.97' Method: OPTIONS node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INFO sip:92.63.138.97:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKfc9e.c4df9706.0 To: sip:92.63.138.97:5060 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-1edc CSeq: 10 INFO Call-ID: 6ad64952-7670@92.63.138.100 Content-Length: 0 User-Agent: OpenSIPS (1.5.1-notls (i386/linux)) <-------------> --- (8 headers 0 lines) --- node4*CLI> <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKfc9e.c4df9706.0;received=92.63.138.100 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-1edc To: sip:92.63.138.97:5060;tag=as3164c463 Call-ID: 6ad64952-7670@92.63.138.100 CSeq: 10 INFO Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INFO sip:92.63.138.97:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKbc9e.7bda6a33.0 To: sip:92.63.138.97:5060 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-a88a CSeq: 10 INFO Call-ID: 6ad64956-7670@92.63.138.100 Content-Length: 0 User-Agent: OpenSIPS (1.5.1-notls (i386/linux)) <-------------> --- (8 headers 0 lines) --- node4*CLI> <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKbc9e.7bda6a33.0;received=92.63.138.100 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-a88a To: sip:92.63.138.97:5060;tag=as02f020d4 Call-ID: 6ad64956-7670@92.63.138.100 CSeq: 10 INFO Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> BYE sip:07837689027@92.63.138.97 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 213.166.5.148;branch=z9hG4bK9a9d.13c3a5d1.0 Via: SIP/2.0/UDP 87.238.72.133:5070;branch=z9hG4bK9a9d.f07d1eb7.0 Via: SIP/2.0/UDP 87.238.72.134;rport=5060;branch=z9hG4bK4j25cg4HNet4j Max-Forwards: 68 From: ;tag=HmXX9KN9evK5e To: "Number-1" ;tag=as77a3588f Call-ID: 503c745d40bb85fe698580104885843a@sipipgw.magrathea.net CSeq: 121001311 BYE User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Sending to 213.166.5.148 : 5060 (no NAT) node4*CLI> <--- Transmitting (no NAT) to 213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 213.166.5.148;branch=z9hG4bK9a9d.13c3a5d1.0;received=213.166.5.148 Via: SIP/2.0/UDP 87.238.72.133:5070;branch=z9hG4bK9a9d.f07d1eb7.0 Via: SIP/2.0/UDP 87.238.72.134;rport=5060;branch=z9hG4bK4j25cg4HNet4j Record-Route: From: ;tag=HmXX9KN9evK5e To: "Number-1" ;tag=as77a3588f Call-ID: 503c745d40bb85fe698580104885843a@sipipgw.magrathea.net CSeq: 121001311 BYE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> -- Executing [h@insight-dialout-external:1] Set("SIP/213.166.5.140-b662a6e0", "CDR(outbound)=02085008811") in new stack node4*CLI> == Spawn extension (insight-dialout-external, 442033933660, 5) exited non-zero on 'SIP/213.166.5.140-b662a6e0' node4*CLI> Scheduling destruction of SIP dialog 'E484B7F4-AC4511DE-9AA2D31F-EFD16753@213.166.5.140' in 32000 ms (Method: ACK) node4*CLI> set_destination: Parsing for address/port to send to node4*CLI> set_destination: set destination to 92.63.138.100, port 5060 Reliably Transmitting (no NAT) to 92.63.138.100:5060: BYE sip:07837689027@213.166.5.140:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK091ac355;rport Route: , Max-Forwards: 70 From: ;tag=as77f20c09 To: "07837689027" ;tag=C66BE99C-269C Call-ID: E484B7F4-AC4511DE-9AA2D31F-EFD16753@213.166.5.140 CSeq: 102 BYE User-Agent: Asterisk PBX 1.6.1.3-rc1 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- node4*CLI> == MixMonitor close filestream node4*CLI> == End MixMonitor Recording SIP/213.166.5.140-b662a6e0 node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK091ac355;rport=5060 From: ;tag=as77f20c09 To: ;tag=C66BE99C-269C Date: Tue, 29 Sep 2009 15:48:16 gmt Call-ID: E484B7F4-AC4511DE-9AA2D31F-EFD16753@213.166.5.140 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 BYE Reason: Q.850;cause=16 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- node4*CLI> SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog 'E484B7F4-AC4511DE-9AA2D31F-EFD16753@213.166.5.140' Method: ACK Really destroying SIP dialog '503c745d40bb85fe698580104885843a@sipipgw.magrathea.net' Method: BYE node4*CLI> Reliably Transmitting (no NAT) to 213.166.5.148:5060: OPTIONS sip:sipipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK4ef29731;rport Max-Forwards: 70 From: "asterisk" ;tag=as3bdac13a To: Contact: Call-ID: 7c1d1357224eb7ad6f47b0f52444052a@92.63.138.97 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.1.3-rc1 Date: Tue, 29 Sep 2009 15:48:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 403 OPTIONS not supported Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK4ef29731;rport=5060 From: "asterisk" ;tag=as3bdac13a To: ;tag=9a264c9a00f926193bf7ce80aab147c3.9210 Call-ID: 7c1d1357224eb7ad6f47b0f52444052a@92.63.138.97 CSeq: 102 OPTIONS Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '7c1d1357224eb7ad6f47b0f52444052a@92.63.138.97' Method: OPTIONS node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INFO sip:92.63.138.97:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK80ae.f2c7ed24.0 To: sip:92.63.138.97:5060 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-16f5 CSeq: 10 INFO Call-ID: 6ad6495a-7670@92.63.138.100 Content-Length: 0 User-Agent: OpenSIPS (1.5.1-notls (i386/linux)) <-------------> --- (8 headers 0 lines) --- <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK80ae.f2c7ed24.0;received=92.63.138.100 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-16f5 To: sip:92.63.138.97:5060;tag=as72877720 Call-ID: 6ad6495a-7670@92.63.138.100 CSeq: 10 INFO Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INVITE sip:448005244731@92.63.138.97:5060 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK882e.c17e4d32.0 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bK882e.1ab36063.0 Via: SIP/2.0/UDP 213.166.5.132:5060;branch=z9hG4bK62E707B1D Remote-Party-ID: ;party=calling;screen=yes;privacy=off From: "07717851570" ;tag=DB93274-1508 To: Date: Tue, 29 Sep 2009 15:48:34 gmt Call-ID: 613C7E7E-AC4611DE-9079E0F5-37CB0CDD@213.166.5.132 Supported: timer,resource-priority,replaces Min-SE: 1800 User-Agent: MSSGW Allow: INVITE, BYE, CANCEL, ACK CSeq: 101 INVITE Max-Forwards: 13 Timestamp: 1254239314 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 418 v=0 o=CiscoSystemsSIP-GW-UserAgent 9546 6532 IN IP4 213.166.5.132 s=SIP Call c=IN IP4 213.166.5.132 t=0 0 m=audio 16656 RTP/AVP 8 18 4 3 98 0 101 c=IN IP4 213.166.5.132 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:4 G723/8000 a=fmtp:4 bitrate=6.3;annexa=no a=rtpmap:3 GSM/8000 a=rtpmap:98 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> node4*CLI> --- (24 headers 17 lines) --- == Using SIP RTP CoS mark 5 Sending to 92.63.138.100 : 5060 (no NAT) Using INVITE request as basis request - 613C7E7E-AC4611DE-9079E0F5-37CB0CDD@213.166.5.132 No matching peer for '07717851570' from '92.63.138.100:5060' Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 3 Found RTP audio format 98 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 213.166.5.132:16656 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Got unsupported a:fmtp in SDP offer Found audio description format G723 for ID 4 Got unsupported a:fmtp in SDP offer Found audio description format GSM for ID 3 Found audio description format G726-32 for ID 98 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8 (alaw), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 213.166.5.132:16656 Looking for 448005244731 in incoming_magrathea (domain 92.63.138.97) node4*CLI> list_route: hop: list_route: hop: node4*CLI> <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK882e.c17e4d32.0;received=92.63.138.100 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bK882e.1ab36063.0 Via: SIP/2.0/UDP 213.166.5.132:5060;branch=z9hG4bK62E707B1D Record-Route: Record-Route: From: "07717851570" ;tag=DB93274-1508 To: Call-ID: 613C7E7E-AC4611DE-9079E0F5-37CB0CDD@213.166.5.132 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> node4*CLI> -- Executing AGI("SIP/213.166.5.132-0a180578", "agi://web0/track.agi?username=0800&campaignName=Google+PPC&campaignId=1459&destination=02082361731&analytics=yes") node4*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERPRES()=allowed) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(accountcode)=0800) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(userfield)=1459) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERID(name)=Google PPC) node4*CLI> -- AGI Script agi://web0/track.agi?username=0800&campaignName=Google+PPC&campaignId=1459&destination=02082361731&analytics=yes completed, returning 0 node4*CLI> -- Executing [448005244731@insight-dialout-external:1] GotoIf("SIP/213.166.5.132-0a180578", "0?2:5") in new stack node4*CLI> -- Goto (insight-dialout-external,448005244731,5) node4*CLI> -- Executing [448005244731@insight-dialout-external:5] Dial("SIP/213.166.5.132-0a180578", "SIP/magrathea-outbound/02082361731,40,Cr") in new stack node4*CLI> == Using SIP RTP CoS mark 5 node4*CLI> Audio is at 92.63.138.97 port 17836 node4*CLI> Adding codec 0x8 (alaw) to SDP node4*CLI> Adding non-codec 0x1 (telephone-event) to SDP node4*CLI> Reliably Transmitting (no NAT) to 213.166.5.148:5060: INVITE sip:02082361731@sipipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK0bb16bbb;rport Max-Forwards: 70 From: "Google PPC" ;tag=as52f3ef11 To: Contact: Call-ID: 50540a1d7e37e1a370c5777d3d385e38@sipipgw.magrathea.net CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "Google PPC" ;privacy=off;screen=yes Date: Tue, 29 Sep 2009 15:48:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 263 v=0 o=root 29367199 29367199 IN IP4 92.63.138.97 s=Asterisk PBX 1.6.1.3-rc1 c=IN IP4 92.63.138.97 t=0 0 m=audio 17836 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- node4*CLI> -- Called magrathea-outbound/02082361731 node4*CLI> <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK882e.c17e4d32.0;received=92.63.138.100 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bK882e.1ab36063.0 Via: SIP/2.0/UDP 213.166.5.132:5060;branch=z9hG4bK62E707B1D Record-Route: Record-Route: From: "07717851570" ;tag=DB93274-1508 To: ;tag=as3c7d7bfb Call-ID: 613C7E7E-AC4611DE-9079E0F5-37CB0CDD@213.166.5.132 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK0bb16bbb;rport=5060 From: "Google PPC" ;tag=as52f3ef11 To: Call-ID: 50540a1d7e37e1a370c5777d3d385e38@sipipgw.magrathea.net CSeq: 102 INVITE Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK0bb16bbb;rport=5060 Record-Route: From: "Google PPC" ;tag=as52f3ef11 To: ;tag=DUyBZ505K3KaH Call-ID: 50540a1d7e37e1a370c5777d3d385e38@sipipgw.magrathea.net CSeq: 102 INVITE Contact: User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Type: application/sdp Content-Length: 181 v=0 o=- 53375511 0 IN IP4 87.238.72.134 s=Cisco SDP 0 c=IN IP4 87.238.72.134 t=0 0 m=audio 41190 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (13 headers 9 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 87.238.72.134:41190 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 87.238.72.134:41190 node4*CLI> -- SIP/magrathea-outbound-0a23f1b0 is making progress passing it to SIP/213.166.5.132-0a180578 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK0bb16bbb;rport=5060 Record-Route: From: "Google PPC" ;tag=as52f3ef11 To: ;tag=DUyBZ505K3KaH Call-ID: 50540a1d7e37e1a370c5777d3d385e38@sipipgw.magrathea.net CSeq: 102 INVITE Contact: User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Min-SE: 120 Content-Type: application/sdp Content-Length: 181 v=0 o=- 53375511 0 IN IP4 87.238.72.134 s=Cisco SDP 0 c=IN IP4 87.238.72.134 t=0 0 m=audio 41190 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (14 headers 9 lines) --- list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.148, port 5060 node4*CLI> Transmitting (no NAT) to 213.166.5.148:5060: ACK sip:87.238.72.134 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK452d41a9;rport Route: Max-Forwards: 70 From: "Google PPC" ;tag=as52f3ef11 To: ;tag=DUyBZ505K3KaH Contact: Call-ID: 50540a1d7e37e1a370c5777d3d385e38@sipipgw.magrathea.net CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "Google PPC" ;privacy=off;screen=yes Content-Length: 0 --- node4*CLI> -- SIP/magrathea-outbound-0a23f1b0 answered SIP/213.166.5.132-0a180578 Audio is at 92.63.138.97 port 19308 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP node4*CLI> <--- Reliably Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK882e.c17e4d32.0;received=92.63.138.100 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bK882e.1ab36063.0 Via: SIP/2.0/UDP 213.166.5.132:5060;branch=z9hG4bK62E707B1D Record-Route: Record-Route: From: "07717851570" ;tag=DB93274-1508 To: ;tag=as3c7d7bfb Call-ID: 613C7E7E-AC4611DE-9079E0F5-37CB0CDD@213.166.5.132 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 267 v=0 o=root 1207034514 1207034514 IN IP4 92.63.138.97 s=Asterisk PBX 1.6.1.3-rc1 c=IN IP4 92.63.138.97 t=0 0 m=audio 19308 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> node4*CLI> -- Packet2Packet bridging SIP/213.166.5.132-0a180578 and SIP/magrathea-outbound-0a23f1b0 node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> ACK sip:448005244731@92.63.138.97:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK882e.c17e4d32.2 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bK882e.1ab36063.2 Via: SIP/2.0/UDP 213.166.5.132:5060;branch=z9hG4bK62E709128D From: ;tag=DB93274-1508 To: ;tag=as3c7d7bfb Date: Tue, 29 Sep 2009 15:48:34 gmt Call-ID: 613C7E7E-AC4611DE-9079E0F5-37CB0CDD@213.166.5.132 Max-Forwards: 13 CSeq: 101 ACK Allow-Events: telephone-event Content-Length: 0 <-------------> --- (13 headers 0 lines) --- node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INFO sip:92.63.138.97:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK40ae.f720b612.0 To: sip:92.63.138.97:5060 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-a0a3 CSeq: 10 INFO Call-ID: 6ad6495e-7670@92.63.138.100 Content-Length: 0 User-Agent: OpenSIPS (1.5.1-notls (i386/linux)) <-------------> --- (8 headers 0 lines) --- <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK40ae.f720b612.0;received=92.63.138.100 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-a0a3 To: sip:92.63.138.97:5060;tag=as7534462d Call-ID: 6ad6495e-7670@92.63.138.100 CSeq: 10 INFO Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INVITE sip:442034118274@92.63.138.97:5060 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKa10a.8f855a63.0 Via: SIP/2.0/UDP 213.166.5.129;branch=z9hG4bKa10a.c5abef53.0 Via: SIP/2.0/UDP 213.166.5.133:5060;branch=z9hG4bK579FC5C1742 From: "07779336505" ;tag=65D81A60-F23 To: Date: Tue, 29 Sep 2009 15:48:59 gmt Call-ID: 6FF853C0-AC4611DE-BF66EFB9-710EBE4B@213.166.5.133 Supported: timer,replaces Min-SE: 1800 User-Agent: MSSGW Allow: INVITE, BYE, CANCEL, ACK CSeq: 101 INVITE Max-Forwards: 13 Remote-Party-ID: ;party=calling;screen=yes;privacy=off Timestamp: 1254239339 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 404 v=0 o=CiscoSystemsSIP-GW-UserAgent 4785 4986 IN IP4 213.166.5.133 s=SIP Call c=IN IP4 213.166.5.133 t=0 0 m=audio 18598 RTP/AVP 8 18 4 3 2 0 101 c=IN IP4 213.166.5.133 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:3 GSM/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (23 headers 17 lines) --- == Using SIP RTP CoS mark 5 Sending to 92.63.138.100 : 5060 (no NAT) Using INVITE request as basis request - 6FF853C0-AC4611DE-BF66EFB9-710EBE4B@213.166.5.133 No matching peer for '07779336505' from '92.63.138.100:5060' Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 3 Found RTP audio format 2 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 213.166.5.133:18598 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Got unsupported a:fmtp in SDP offer Found audio description format G723 for ID 4 Got unsupported a:fmtp in SDP offer Found audio description format GSM for ID 3 Found audio description format G726-32 for ID 2 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8 (alaw), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 213.166.5.133:18598 Looking for 442034118274 in incoming_magrathea (domain 92.63.138.97) node4*CLI> list_route: hop: list_route: hop: <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKa10a.8f855a63.0;received=92.63.138.100 Via: SIP/2.0/UDP 213.166.5.129;branch=z9hG4bKa10a.c5abef53.0 Via: SIP/2.0/UDP 213.166.5.133:5060;branch=z9hG4bK579FC5C1742 Record-Route: Record-Route: From: "07779336505" ;tag=65D81A60-F23 To: Call-ID: 6FF853C0-AC4611DE-BF66EFB9-710EBE4B@213.166.5.133 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> node4*CLI> -- Executing AGI("SIP/213.166.5.133-b662b550", "agi://web0/track.agi?username=provisioning&campaignName=Number-2&campaignId=824&destination=02072266066&recordCall=yes") node4*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERPRES()=allowed) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(accountcode)=provisioning) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(userfield)=824) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERID(name)=Number-2) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(amaflags)=billing) node4*CLI> -- AGI Script agi://web0/track.agi?username=provisioning&campaignName=Number-2&campaignId=824&destination=02072266066&recordCall=yes completed, returning 0 node4*CLI> -- Executing [442034118274@insight-dialout-external:1] GotoIf("SIP/213.166.5.133-b662b550", "1?2:5") in new stack -- Goto (insight-dialout-external,442034118274,2) -- Executing [442034118274@insight-dialout-external:2] Answer("SIP/213.166.5.133-b662b550", "") in new stack node4*CLI> Audio is at 92.63.138.97 port 14358 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKa10a.8f855a63.0;received=92.63.138.100 Via: SIP/2.0/UDP 213.166.5.129;branch=z9hG4bKa10a.c5abef53.0 Via: SIP/2.0/UDP 213.166.5.133:5060;branch=z9hG4bK579FC5C1742 Record-Route: Record-Route: From: "07779336505" ;tag=65D81A60-F23 To: ;tag=as500f55d1 Call-ID: 6FF853C0-AC4611DE-BF66EFB9-710EBE4B@213.166.5.133 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 265 v=0 o=root 757301666 757301666 IN IP4 92.63.138.97 s=Asterisk PBX 1.6.1.3-rc1 c=IN IP4 92.63.138.97 t=0 0 m=audio 14358 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> ACK sip:442034118274@92.63.138.97:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKa10a.8f855a63.2 Via: SIP/2.0/UDP 213.166.5.129;branch=z9hG4bKa10a.c5abef53.2 Via: SIP/2.0/UDP 213.166.5.133:5060;branch=z9hG4bK579FC5D298 From: ;tag=65D81A60-F23 To: ;tag=as500f55d1 Date: Tue, 29 Sep 2009 15:48:59 gmt Call-ID: 6FF853C0-AC4611DE-BF66EFB9-710EBE4B@213.166.5.133 Max-Forwards: 13 CSeq: 101 ACK Content-Length: 0 <-------------> --- (12 headers 0 lines) --- node4*CLI> -- Executing [442034118274@insight-dialout-external:3] MixMonitor("SIP/213.166.5.133-b662b550", "voip2-1254239339.645.wav") in new stack node4*CLI> -- Executing [442034118274@insight-dialout-external:4] BackGround("SIP/213.166.5.133-b662b550", "adinsight-call-recorded") in new stack == Begin MixMonitor Recording SIP/213.166.5.133-b662b550 node4*CLI> -- Playing 'adinsight-call-recorded.ulaw' (language 'en') node4*CLI> -- Executing [442034118274@insight-dialout-external:5] Dial("SIP/213.166.5.133-b662b550", "SIP/magrathea-outbound/02072266066,40,Cr") in new stack node4*CLI> == Using SIP RTP CoS mark 5 node4*CLI> Audio is at 92.63.138.97 port 19856 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP node4*CLI> Reliably Transmitting (no NAT) to 213.166.5.148:5060: INVITE sip:02072266066@sipipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK0e36c129;rport Max-Forwards: 70 From: "Number-2" ;tag=as5a444872 To: Contact: Call-ID: 76b7bf8824076b182f31c06f02ee4ff7@sipipgw.magrathea.net CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "Number-2" ;privacy=off;screen=yes Date: Tue, 29 Sep 2009 15:49:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 265 v=0 o=root 626144779 626144779 IN IP4 92.63.138.97 s=Asterisk PBX 1.6.1.3-rc1 c=IN IP4 92.63.138.97 t=0 0 m=audio 19856 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- node4*CLI> -- Called magrathea-outbound/02072266066 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK0e36c129;rport=5060 From: "Number-2" ;tag=as5a444872 To: Call-ID: 76b7bf8824076b182f31c06f02ee4ff7@sipipgw.magrathea.net CSeq: 102 INVITE Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK0e36c129;rport=5060 Record-Route: From: "Number-2" ;tag=as5a444872 To: ;tag=437gjpQmQjr7D Call-ID: 76b7bf8824076b182f31c06f02ee4ff7@sipipgw.magrathea.net CSeq: 102 INVITE Contact: User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Type: application/sdp Content-Length: 181 v=0 o=- 53375854 0 IN IP4 213.166.5.153 s=Cisco SDP 0 c=IN IP4 213.166.5.153 t=0 0 m=audio 34970 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (13 headers 9 lines) --- node4*CLI> Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 213.166.5.153:34970 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 213.166.5.153:34970 node4*CLI> -- SIP/magrathea-outbound-0a1ecf08 is making progress passing it to SIP/213.166.5.133-b662b550 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK0e36c129;rport=5060 Record-Route: From: "Number-2" ;tag=as5a444872 To: ;tag=437gjpQmQjr7D Call-ID: 76b7bf8824076b182f31c06f02ee4ff7@sipipgw.magrathea.net CSeq: 102 INVITE Contact: User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Min-SE: 120 Content-Type: application/sdp Content-Length: 181 v=0 o=- 53375854 0 IN IP4 213.166.5.153 s=Cisco SDP 0 c=IN IP4 213.166.5.153 t=0 0 m=audio 34970 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (14 headers 9 lines) --- list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.148, port 5060 Transmitting (no NAT) to 213.166.5.148:5060: ACK sip:213.166.5.153 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK42a5fee3;rport Route: Max-Forwards: 70 From: "Number-2" ;tag=as5a444872 To: ;tag=437gjpQmQjr7D Contact: Call-ID: 76b7bf8824076b182f31c06f02ee4ff7@sipipgw.magrathea.net CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "Number-2" ;privacy=off;screen=yes Content-Length: 0 --- node4*CLI> -- SIP/magrathea-outbound-0a1ecf08 answered SIP/213.166.5.133-b662b550 node4*CLI> Reliably Transmitting (no NAT) to 213.166.5.148:5060: OPTIONS sip:sipipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK5bc60561;rport Max-Forwards: 70 From: "asterisk" ;tag=as579cc115 To: Contact: Call-ID: 160195856f5e85a809fd3e2216700071@92.63.138.97 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.1.3-rc1 Date: Tue, 29 Sep 2009 15:49:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 403 OPTIONS not supported Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK5bc60561;rport=5060 From: "asterisk" ;tag=as579cc115 To: ;tag=9a264c9a00f926193bf7ce80aab147c3.a7c4 Call-ID: 160195856f5e85a809fd3e2216700071@92.63.138.97 CSeq: 102 OPTIONS Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- node4*CLI> Really destroying SIP dialog '160195856f5e85a809fd3e2216700071@92.63.138.97' Method: OPTIONS node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INFO sip:92.63.138.97:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKf27e.904e243.0 To: sip:92.63.138.97:5060 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-5549 CSeq: 10 INFO Call-ID: 6ad64962-7670@92.63.138.100 Content-Length: 0 User-Agent: OpenSIPS (1.5.1-notls (i386/linux)) <-------------> --- (8 headers 0 lines) --- <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKf27e.904e243.0;received=92.63.138.100 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-5549 To: sip:92.63.138.97:5060;tag=as210fdf93 Call-ID: 6ad64962-7670@92.63.138.100 CSeq: 10 INFO Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> BYE sip:442034118212@92.63.138.97:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK71f.159f7784.0 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bK71f.193591f7.0 Via: SIP/2.0/UDP 213.166.5.140:5060;branch=z9hG4bK5044561D92 From: "anonymous" ;tag=C66C5284-208F To: ;tag=as253f7d2b Date: Tue, 29 Sep 2009 15:45:32 gmt Call-ID: F486A1D5-AC4511DE-9BD6D31F-EFD16753@213.166.5.140 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 13 Timestamp: 1254239367 CSeq: 102 BYE Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Sending to 92.63.138.100 : 5060 (no NAT) <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK71f.159f7784.0;received=92.63.138.100 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bK71f.193591f7.0 Via: SIP/2.0/UDP 213.166.5.140:5060;branch=z9hG4bK5044561D92 Record-Route: From: "anonymous" ;tag=C66C5284-208F To: ;tag=as253f7d2b Call-ID: F486A1D5-AC4511DE-9BD6D31F-EFD16753@213.166.5.140 CSeq: 102 BYE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> -- Executing [h@insight-dialout-external:1] Set("SIP/213.166.5.140-b662fa10", "CDR(outbound)=01708443551") in new stack node4*CLI> Scheduling destruction of SIP dialog '55c5cf6d6d89fec63b61df7e63ee0163@sipipgw.magrathea.net' in 6400 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.148, port 5060 Reliably Transmitting (no NAT) to 213.166.5.148:5060: BYE sip:87.238.72.134 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK2f2dbeb6;rport Route: Max-Forwards: 70 From: "Number-1" ;tag=as17d751b6 To: ;tag=9r8ZXje8KFNHN Call-ID: 55c5cf6d6d89fec63b61df7e63ee0163@sipipgw.magrathea.net CSeq: 103 BYE User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "Number-1" ;privacy=off;screen=yes X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- == Spawn extension (insight-dialout-external, 442034118212, 5) exited non-zero on 'SIP/213.166.5.140-b662fa10' node4*CLI> == MixMonitor close filestream == End MixMonitor Recording SIP/213.166.5.140-b662fa10 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK2f2dbeb6;rport=5060 From: "Number-1" ;tag=as17d751b6 To: ;tag=9r8ZXje8KFNHN Call-ID: 55c5cf6d6d89fec63b61df7e63ee0163@sipipgw.magrathea.net CSeq: 103 BYE User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Length: 0 <-------------> --- (10 headers 0 lines) --- node4*CLI> Really destroying SIP dialog '55c5cf6d6d89fec63b61df7e63ee0163@sipipgw.magrathea.net' Method: INVITE Really destroying SIP dialog 'F486A1D5-AC4511DE-9BD6D31F-EFD16753@213.166.5.140' Method: BYE node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> BYE sip:01707333573@92.63.138.97:5060 SIP/2.0 Via: SIP/2.0/UDP 213.166.5.148;branch=z9hG4bKf10d.a46aae84.0 Via: SIP/2.0/UDP 87.238.72.155:5060;branch=z9hG4bK1FD04415A1 From: ;tag=A5CE517C-176E To: "Number-1" ;tag=as21cb1aeb Date: Tue, 29 Sep 2009 15:44:39 gmt Call-ID: 3db3c73d482e72f56170316078909b5f@sipipgw.magrathea.net User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 14 Timestamp: 1254239383 CSeq: 101 BYE Reason: Q.850;cause=16 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 213.166.5.148 : 5060 (no NAT) node4*CLI> <--- Transmitting (no NAT) to 213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 213.166.5.148;branch=z9hG4bKf10d.a46aae84.0;received=213.166.5.148 Via: SIP/2.0/UDP 87.238.72.155:5060;branch=z9hG4bK1FD04415A1 From: ;tag=A5CE517C-176E To: "Number-1" ;tag=as21cb1aeb Call-ID: 3db3c73d482e72f56170316078909b5f@sipipgw.magrathea.net CSeq: 101 BYE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> -- Executing [h@insight-dialout-external:1] Set("SIP/213.166.5.133-0a18cec0", "CDR(outbound)=44800289752") in new stack node4*CLI> == Spawn extension (insight-dialout-external, 448081201315, 5) exited non-zero on 'SIP/213.166.5.133-0a18cec0' node4*CLI> Scheduling destruction of SIP dialog 'CAFB850B-AC4511DE-B259EFB9-710EBE4B@213.166.5.133' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 92.63.138.100, port 5060 Reliably Transmitting (no NAT) to 92.63.138.100:5060: BYE sip:01707333573@213.166.5.133:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK4665da68;rport Route: , Max-Forwards: 70 From: ;tag=as3cf2c569 To: "01707333573" ;tag=65D3E11C-12ED Call-ID: CAFB850B-AC4511DE-B259EFB9-710EBE4B@213.166.5.133 CSeq: 102 BYE User-Agent: Asterisk PBX 1.6.1.3-rc1 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- node4*CLI> == MixMonitor close filestream node4*CLI> == End MixMonitor Recording SIP/213.166.5.133-0a18cec0 node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK4665da68;rport=5060 From: ;tag=as3cf2c569 To: ;tag=65D3E11C-12ED Date: Tue, 29 Sep 2009 15:49:43 gmt Call-ID: CAFB850B-AC4511DE-B259EFB9-710EBE4B@213.166.5.133 Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 102 BYE <-------------> --- (9 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog 'CAFB850B-AC4511DE-B259EFB9-710EBE4B@213.166.5.133' Method: ACK Really destroying SIP dialog '3db3c73d482e72f56170316078909b5f@sipipgw.magrathea.net' Method: BYE node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INFO sip:92.63.138.97:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKb27e.2843e1a1.0 To: sip:92.63.138.97:5060 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-e31f CSeq: 10 INFO Call-ID: 6ad64966-7670@92.63.138.100 Content-Length: 0 User-Agent: OpenSIPS (1.5.1-notls (i386/linux)) <-------------> --- (8 headers 0 lines) --- <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKb27e.2843e1a1.0;received=92.63.138.100 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-e31f To: sip:92.63.138.97:5060;tag=as6841f280 Call-ID: 6ad64966-7670@92.63.138.100 CSeq: 10 INFO Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INVITE sip:448005244040@92.63.138.97:5060 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK8306.e2403787.0 Via: SIP/2.0/UDP 87.238.72.153;branch=z9hG4bK8306.5e87a6f3.0 Via: SIP/2.0/UDP 87.238.72.149:5060;branch=z9hG4bK1694DCD13BC From: "01619734347" ;tag=B0D96F1C-159D To: Date: Tue, 29 Sep 2009 15:49:56 gmt Call-ID: 92398592-AC4611DE-BF0EF3EC-65114225@87.238.72.149 Supported: timer,replaces Min-SE: 1800 User-Agent: MSSGW(B) Allow: INVITE, BYE, CANCEL, ACK CSeq: 101 INVITE Max-Forwards: 13 Remote-Party-ID: ;party=calling;screen=yes;privacy=off Timestamp: 1254239396 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 404 v=0 o=CiscoSystemsSIP-GW-UserAgent 8072 8078 IN IP4 87.238.72.149 s=SIP Call c=IN IP4 87.238.72.149 t=0 0 m=audio 18978 RTP/AVP 8 18 4 3 2 0 101 c=IN IP4 87.238.72.149 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:3 GSM/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (23 headers 17 lines) --- == Using SIP RTP CoS mark 5 Sending to 92.63.138.100 : 5060 (no NAT) Using INVITE request as basis request - 92398592-AC4611DE-BF0EF3EC-65114225@87.238.72.149 No matching peer for '01619734347' from '92.63.138.100:5060' Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 3 Found RTP audio format 2 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 87.238.72.149:18978 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Got unsupported a:fmtp in SDP offer node4*CLI> Found audio description format G723 for ID 4 Got unsupported a:fmtp in SDP offer Found audio description format GSM for ID 3 Found audio description format G726-32 for ID 2 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8 (alaw), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 87.238.72.149:18978 Looking for 448005244040 in incoming_magrathea (domain 92.63.138.97) node4*CLI> list_route: hop: list_route: hop: <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK8306.e2403787.0;received=92.63.138.100 Via: SIP/2.0/UDP 87.238.72.153;branch=z9hG4bK8306.5e87a6f3.0 Via: SIP/2.0/UDP 87.238.72.149:5060;branch=z9hG4bK1694DCD13BC Record-Route: Record-Route: From: "01619734347" ;tag=B0D96F1C-159D To: Call-ID: 92398592-AC4611DE-BF0EF3EC-65114225@87.238.72.149 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> node4*CLI> -- Executing AGI("SIP/87.238.72.149-b664a630", "agi://web0/track.agi?username=searchofficespace&campaignName=direct+traffic&campaignId=1255&destination=02089095222&salesTracking=yes&recordCall=yes&analytics=yes") node4*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERPRES()=allowed) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(accountcode)=searchofficespace) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(userfield)=1255) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERID(name)=direct traffic) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(amaflags)=billing) node4*CLI> -- AGI Script agi://web0/track.agi?username=searchofficespace&campaignName=direct+traffic&campaignId=1255&destination=02089095222&salesTracking=yes&recordCall=yes&analytics=yes completed, returning 0 node4*CLI> -- Executing [448005244040@insight-dialout-external:1] GotoIf("SIP/87.238.72.149-b664a630", "1?2:5") in new stack node4*CLI> -- Goto (insight-dialout-external,448005244040,2) node4*CLI> -- Executing [448005244040@insight-dialout-external:2] Answer("SIP/87.238.72.149-b664a630", "") in new stack node4*CLI> Audio is at 92.63.138.97 port 11642 node4*CLI> Adding codec 0x8 (alaw) to SDP node4*CLI> Adding non-codec 0x1 (telephone-event) to SDP node4*CLI> <--- Reliably Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK8306.e2403787.0;received=92.63.138.100 Via: SIP/2.0/UDP 87.238.72.153;branch=z9hG4bK8306.5e87a6f3.0 Via: SIP/2.0/UDP 87.238.72.149:5060;branch=z9hG4bK1694DCD13BC Record-Route: Record-Route: From: "01619734347" ;tag=B0D96F1C-159D To: ;tag=as2d71f2b0 Call-ID: 92398592-AC4611DE-BF0EF3EC-65114225@87.238.72.149 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 81041556 81041556 IN IP4 92.63.138.97 s=Asterisk PBX 1.6.1.3-rc1 c=IN IP4 92.63.138.97 t=0 0 m=audio 11642 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> ACK sip:448005244040@92.63.138.97:5060 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK8306.e2403787.2 Via: SIP/2.0/UDP 87.238.72.153;branch=z9hG4bK8306.5e87a6f3.2 Via: SIP/2.0/UDP 87.238.72.149:5060;branch=z9hG4bK1694DCE1763 From: ;tag=B0D96F1C-159D To: ;tag=as2d71f2b0 Date: Tue, 29 Sep 2009 15:49:56 gmt Call-ID: 92398592-AC4611DE-BF0EF3EC-65114225@87.238.72.149 Max-Forwards: 13 CSeq: 101 ACK Content-Length: 0 <-------------> --- (13 headers 0 lines) --- node4*CLI> -- Executing [448005244040@insight-dialout-external:3] MixMonitor("SIP/87.238.72.149-b664a630", "voip2-1254239396.647.wav") in new stack node4*CLI> == Begin MixMonitor Recording SIP/87.238.72.149-b664a630 node4*CLI> -- Executing [448005244040@insight-dialout-external:4] BackGround("SIP/87.238.72.149-b664a630", "adinsight-call-recorded") in new stack node4*CLI> -- Playing 'adinsight-call-recorded.ulaw' (language 'en') node4*CLI> -- Executing [448005244040@insight-dialout-external:5] Dial("SIP/87.238.72.149-b664a630", "SIP/magrathea-outbound/02089095222,40,CrF(trackSales^448005244040:1255:voip2-1254239396.647:02089095222^1)") in new stack node4*CLI> == Using SIP RTP CoS mark 5 node4*CLI> Audio is at 92.63.138.97 port 15824 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP node4*CLI> Reliably Transmitting (no NAT) to 213.166.5.148:5060: INVITE sip:02089095222@sipipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK64fa3676;rport Max-Forwards: 70 From: "direct traffic" ;tag=as5d938583 To: Contact: Call-ID: 19d637887454386e0f55e269610cd8ad@sipipgw.magrathea.net CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "direct traffic" ;privacy=off;screen=yes Date: Tue, 29 Sep 2009 15:50:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 265 v=0 o=root 925406982 925406982 IN IP4 92.63.138.97 s=Asterisk PBX 1.6.1.3-rc1 c=IN IP4 92.63.138.97 t=0 0 m=audio 15824 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called magrathea-outbound/02089095222 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK64fa3676;rport=5060 From: "direct traffic" ;tag=as5d938583 To: Call-ID: 19d637887454386e0f55e269610cd8ad@sipipgw.magrathea.net CSeq: 102 INVITE Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK64fa3676;rport=5060 Record-Route: From: "direct traffic" ;tag=as5d938583 To: ;tag=ZHXZ6yX94QS3c Call-ID: 19d637887454386e0f55e269610cd8ad@sipipgw.magrathea.net CSeq: 102 INVITE Contact: User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Type: application/sdp Content-Length: 181 v=0 o=- 53376467 0 IN IP4 213.166.5.153 s=Cisco SDP 0 c=IN IP4 213.166.5.153 t=0 0 m=audio 36110 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (13 headers 9 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 213.166.5.153:36110 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 213.166.5.153:36110 -- SIP/magrathea-outbound-0a18cec0 is making progress passing it to SIP/87.238.72.149-b664a630 node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> BYE sip:448005244731@92.63.138.97:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK582e.b5b8fd22.0 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bK582e.dfa0fe5.0 Via: SIP/2.0/UDP 213.166.5.132:5060;branch=z9hG4bK62E761684 From: ;tag=DB93274-1508 To: ;tag=as3c7d7bfb Date: Tue, 29 Sep 2009 15:48:34 gmt Call-ID: 613C7E7E-AC4611DE-9079E0F5-37CB0CDD@213.166.5.132 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 13 Timestamp: 1254239402 CSeq: 102 BYE Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Sending to 92.63.138.100 : 5060 (no NAT) <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK582e.b5b8fd22.0;received=92.63.138.100 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bK582e.dfa0fe5.0 Via: SIP/2.0/UDP 213.166.5.132:5060;branch=z9hG4bK62E761684 Record-Route: From: ;tag=DB93274-1508 To: ;tag=as3c7d7bfb Call-ID: 613C7E7E-AC4611DE-9079E0F5-37CB0CDD@213.166.5.132 CSeq: 102 BYE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> -- Executing [h@insight-dialout-external:1] Set("SIP/213.166.5.132-0a180578", "CDR(outbound)=02082361731") in new stack node4*CLI> Scheduling destruction of SIP dialog '50540a1d7e37e1a370c5777d3d385e38@sipipgw.magrathea.net' in 6400 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.148, port 5060 Reliably Transmitting (no NAT) to 213.166.5.148:5060: BYE sip:87.238.72.134 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK4bc9da45;rport Route: Max-Forwards: 70 From: "Google PPC" ;tag=as52f3ef11 To: ;tag=DUyBZ505K3KaH Call-ID: 50540a1d7e37e1a370c5777d3d385e38@sipipgw.magrathea.net CSeq: 103 BYE User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "Google PPC" ;privacy=off;screen=yes X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- == Spawn extension (insight-dialout-external, 448005244731, 5) exited non-zero on 'SIP/213.166.5.132-0a180578' node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK4bc9da45;rport=5060 From: "Google PPC" ;tag=as52f3ef11 To: ;tag=DUyBZ505K3KaH Call-ID: 50540a1d7e37e1a370c5777d3d385e38@sipipgw.magrathea.net CSeq: 103 BYE User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '613C7E7E-AC4611DE-9079E0F5-37CB0CDD@213.166.5.132' Method: BYE Really destroying SIP dialog '50540a1d7e37e1a370c5777d3d385e38@sipipgw.magrathea.net' Method: INVITE node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INVITE sip:442033936468@92.63.138.97:5060 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK78b3.247d1a54.0 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bK78b3.f6bba033.0 Via: SIP/2.0/UDP 213.166.5.140:5060;branch=z9hG4bK5044D0307 Remote-Party-ID: ;party=calling;screen=yes;privacy=off From: "02088462000" ;tag=C67080C4-206A To: Date: Tue, 29 Sep 2009 15:50:06 gmt Call-ID: 97D5636A-AC4611DE-A8D9D31F-EFD16753@213.166.5.140 Supported: timer,resource-priority,replaces Min-SE: 1800 User-Agent: MSSGW Allow: INVITE, BYE, CANCEL, ACK CSeq: 101 INVITE Max-Forwards: 13 Timestamp: 1254239406 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 418 v=0 o=CiscoSystemsSIP-GW-UserAgent 1689 6214 IN IP4 213.166.5.140 s=SIP Call c=IN IP4 213.166.5.140 t=0 0 m=audio 20596 RTP/AVP 8 18 4 3 98 0 101 c=IN IP4 213.166.5.140 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:4 G723/8000 a=fmtp:4 bitrate=6.3;annexa=no a=rtpmap:3 GSM/8000 a=rtpmap:98 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (24 headers 17 lines) --- == Using SIP RTP CoS mark 5 Sending to 92.63.138.100 : 5060 (no NAT) Using INVITE request as basis request - 97D5636A-AC4611DE-A8D9D31F-EFD16753@213.166.5.140 No matching peer for '02088462000' from '92.63.138.100:5060' Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 3 Found RTP audio format 98 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 213.166.5.140:20596 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Got unsupported a:fmtp in SDP offer Found audio description format G723 for ID 4 Got unsupported a:fmtp in SDP offer Found audio description format GSM for ID 3 Found audio description format G726-32 for ID 98 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8 (alaw), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 213.166.5.140:20596 Looking for 442033936468 in incoming_magrathea (domain 92.63.138.97) node4*CLI> list_route: hop: list_route: hop: <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK78b3.247d1a54.0;received=92.63.138.100 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bK78b3.f6bba033.0 Via: SIP/2.0/UDP 213.166.5.140:5060;branch=z9hG4bK5044D0307 Record-Route: Record-Route: From: "02088462000" ;tag=C67080C4-206A To: Call-ID: 97D5636A-AC4611DE-A8D9D31F-EFD16753@213.166.5.140 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> node4*CLI> -- Executing AGI("SIP/213.166.5.140-b660f528", "agi://web0/track.agi?username=provisioning&campaignName=Number-5&campaignId=757&destination=02077360193&recordCall=yes") node4*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERPRES()=allowed) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(accountcode)=provisioning) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(userfield)=757) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERID(name)=Number-5) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(amaflags)=billing) node4*CLI> -- AGI Script agi://web0/track.agi?username=provisioning&campaignName=Number-5&campaignId=757&destination=02077360193&recordCall=yes completed, returning 0 node4*CLI> -- Executing [442033936468@insight-dialout-external:1] GotoIf("SIP/213.166.5.140-b660f528", "1?2:5") in new stack node4*CLI> -- Goto (insight-dialout-external,442033936468,2) node4*CLI> -- Executing [442033936468@insight-dialout-external:2] Answer("SIP/213.166.5.140-b660f528", "") in new stack node4*CLI> Audio is at 92.63.138.97 port 14978 node4*CLI> Adding codec 0x8 (alaw) to SDP node4*CLI> Adding non-codec 0x1 (telephone-event) to SDP node4*CLI> <--- Reliably Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK78b3.247d1a54.0;received=92.63.138.100 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bK78b3.f6bba033.0 Via: SIP/2.0/UDP 213.166.5.140:5060;branch=z9hG4bK5044D0307 Record-Route: Record-Route: From: "02088462000" ;tag=C67080C4-206A To: ;tag=as2398868e Call-ID: 97D5636A-AC4611DE-A8D9D31F-EFD16753@213.166.5.140 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 267 v=0 o=root 1794040809 1794040809 IN IP4 92.63.138.97 s=Asterisk PBX 1.6.1.3-rc1 c=IN IP4 92.63.138.97 t=0 0 m=audio 14978 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> ACK sip:442033936468@92.63.138.97:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK78b3.247d1a54.2 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bK78b3.f6bba033.2 Via: SIP/2.0/UDP 213.166.5.140:5060;branch=z9hG4bK5044D184D From: ;tag=C67080C4-206A To: ;tag=as2398868e Date: Tue, 29 Sep 2009 15:50:06 gmt Call-ID: 97D5636A-AC4611DE-A8D9D31F-EFD16753@213.166.5.140 Max-Forwards: 13 CSeq: 101 ACK Allow-Events: telephone-event Content-Length: 0 <-------------> --- (13 headers 0 lines) --- node4*CLI> -- Executing [442033936468@insight-dialout-external:3] MixMonitor("SIP/213.166.5.140-b660f528", "voip2-1254239406.649.wav") in new stack node4*CLI> -- Executing [442033936468@insight-dialout-external:4] BackGround("SIP/213.166.5.140-b660f528", "adinsight-call-recorded") in new stack == Begin MixMonitor Recording SIP/213.166.5.140-b660f528 node4*CLI> -- Playing 'adinsight-call-recorded.ulaw' (language 'en') node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK64fa3676;rport=5060 Record-Route: From: "direct traffic" ;tag=as5d938583 To: ;tag=ZHXZ6yX94QS3c Call-ID: 19d637887454386e0f55e269610cd8ad@sipipgw.magrathea.net CSeq: 102 INVITE Contact: User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Min-SE: 120 Content-Type: application/sdp Content-Length: 181 v=0 o=- 53376467 0 IN IP4 213.166.5.153 s=Cisco SDP 0 c=IN IP4 213.166.5.153 t=0 0 m=audio 36110 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (14 headers 9 lines) --- list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.148, port 5060 Transmitting (no NAT) to 213.166.5.148:5060: ACK sip:213.166.5.153 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK4709c7ae;rport Route: Max-Forwards: 70 From: "direct traffic" ;tag=as5d938583 To: ;tag=ZHXZ6yX94QS3c Contact: Call-ID: 19d637887454386e0f55e269610cd8ad@sipipgw.magrathea.net CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "direct traffic" ;privacy=off;screen=yes Content-Length: 0 --- node4*CLI> -- SIP/magrathea-outbound-0a18cec0 answered SIP/87.238.72.149-b664a630 node4*CLI> -- Executing [442033936468@insight-dialout-external:5] Dial("SIP/213.166.5.140-b660f528", "SIP/magrathea-outbound/02077360193,40,Cr") in new stack node4*CLI> == Using SIP RTP CoS mark 5 node4*CLI> Audio is at 92.63.138.97 port 10512 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP node4*CLI> Reliably Transmitting (no NAT) to 213.166.5.148:5060: INVITE sip:02077360193@sipipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK7ddcbe1b;rport Max-Forwards: 70 From: "Number-5" ;tag=as7d9d02d7 To: Contact: Call-ID: 07f24a2a3502e10b0335f3e3628d310e@sipipgw.magrathea.net CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "Number-5" ;privacy=off;screen=yes Date: Tue, 29 Sep 2009 15:50:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 265 v=0 o=root 956833918 956833918 IN IP4 92.63.138.97 s=Asterisk PBX 1.6.1.3-rc1 c=IN IP4 92.63.138.97 t=0 0 m=audio 10512 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called magrathea-outbound/02077360193 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK7ddcbe1b;rport=5060 From: "Number-5" ;tag=as7d9d02d7 To: Call-ID: 07f24a2a3502e10b0335f3e3628d310e@sipipgw.magrathea.net CSeq: 102 INVITE Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK7ddcbe1b;rport=5060 Record-Route: From: "Number-5" ;tag=as7d9d02d7 To: ;tag=eK08Dm90p4cmB Call-ID: 07f24a2a3502e10b0335f3e3628d310e@sipipgw.magrathea.net CSeq: 102 INVITE Contact: User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Type: application/sdp Content-Length: 181 v=0 o=- 53376586 0 IN IP4 87.238.72.134 s=Cisco SDP 0 c=IN IP4 87.238.72.134 t=0 0 m=audio 43154 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (13 headers 9 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 87.238.72.134:43154 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 87.238.72.134:43154 node4*CLI> -- SIP/magrathea-outbound-b6608498 is making progress passing it to SIP/213.166.5.140-b660f528 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK7ddcbe1b;rport=5060 Record-Route: From: "Number-5" ;tag=as7d9d02d7 To: ;tag=eK08Dm90p4cmB Call-ID: 07f24a2a3502e10b0335f3e3628d310e@sipipgw.magrathea.net CSeq: 102 INVITE Contact: User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Min-SE: 120 Content-Type: application/sdp Content-Length: 181 v=0 o=- 53376586 0 IN IP4 87.238.72.134 s=Cisco SDP 0 c=IN IP4 87.238.72.134 t=0 0 m=audio 43154 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (14 headers 9 lines) --- list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.148, port 5060 Transmitting (no NAT) to 213.166.5.148:5060: ACK sip:87.238.72.134 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK46ec3ae8;rport Route: Max-Forwards: 70 From: "Number-5" ;tag=as7d9d02d7 To: ;tag=eK08Dm90p4cmB Contact: Call-ID: 07f24a2a3502e10b0335f3e3628d310e@sipipgw.magrathea.net CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "Number-5" ;privacy=off;screen=yes Content-Length: 0 --- node4*CLI> -- SIP/magrathea-outbound-b6608498 answered SIP/213.166.5.140-b660f528 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> BYE sip:01455554861@92.63.138.97 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 213.166.5.148;branch=z9hG4bK146e.3138ac47.0 Via: SIP/2.0/UDP 213.166.5.139:5070;branch=z9hG4bK146e.7ae0fef5.0 Via: SIP/2.0/UDP 213.166.5.153;rport=5060;branch=z9hG4bKyUQaNNHa6yDHQ Max-Forwards: 68 From: ;tag=gtevK2ap5N5BH To: "freestart main number" ;tag=as0c6dbab3 Call-ID: 3c8996672e1cc9f330e9a6cb27130d09@sipipgw.magrathea.net CSeq: 121001372 BYE User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Length: 0 <-------------> --- (14 headers 0 lines) --- node4*CLI> Sending to 213.166.5.148 : 5060 (no NAT) <--- Transmitting (no NAT) to 213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 213.166.5.148;branch=z9hG4bK146e.3138ac47.0;received=213.166.5.148 Via: SIP/2.0/UDP 213.166.5.139:5070;branch=z9hG4bK146e.7ae0fef5.0 Via: SIP/2.0/UDP 213.166.5.153;rport=5060;branch=z9hG4bKyUQaNNHa6yDHQ Record-Route: From: ;tag=gtevK2ap5N5BH To: "freestart main number" ;tag=as0c6dbab3 Call-ID: 3c8996672e1cc9f330e9a6cb27130d09@sipipgw.magrathea.net CSeq: 121001372 BYE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> -- Executing [h@insight-dialout-external:1] Set("SIP/87.238.72.149-b7b87118", "CDR(outbound)=01942406100") in new stack node4*CLI> == Spawn extension (insight-dialout-external, 448447747777, 5) exited non-zero on 'SIP/87.238.72.149-b7b87118' node4*CLI> Scheduling destruction of SIP dialog 'F4E5DAE-AC4611DE-BB83F3EC-65114225@87.238.72.149' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 92.63.138.100, port 5060 Reliably Transmitting (no NAT) to 92.63.138.100:5060: BYE sip:01455554861@87.238.72.149:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK5ee3c1d9;rport Route: , Max-Forwards: 70 From: ;tag=as2661c4ae To: "01455554861" ;tag=B0D61520-1BBC Call-ID: F4E5DAE-AC4611DE-BB83F3EC-65114225@87.238.72.149 CSeq: 102 BYE User-Agent: Asterisk PBX 1.6.1.3-rc1 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- node4*CLI> == MixMonitor close filestream == End MixMonitor Recording SIP/87.238.72.149-b7b87118 node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK5ee3c1d9;rport=5060 From: ;tag=as2661c4ae To: ;tag=B0D61520-1BBC Date: Tue, 29 Sep 2009 15:50:18 gmt Call-ID: F4E5DAE-AC4611DE-BB83F3EC-65114225@87.238.72.149 Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 102 BYE <-------------> --- (9 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog 'F4E5DAE-AC4611DE-BB83F3EC-65114225@87.238.72.149' Method: ACK Really destroying SIP dialog '3c8996672e1cc9f330e9a6cb27130d09@sipipgw.magrathea.net' Method: BYE node4*CLI> Reliably Transmitting (no NAT) to 213.166.5.148:5060: OPTIONS sip:sipipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK259b2b06;rport Max-Forwards: 70 From: "asterisk" ;tag=as5cf873fc To: Contact: Call-ID: 686706001c38b1837c7071c84761b300@92.63.138.97 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.1.3-rc1 Date: Tue, 29 Sep 2009 15:50:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 403 OPTIONS not supported Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK259b2b06;rport=5060 From: "asterisk" ;tag=as5cf873fc To: ;tag=9a264c9a00f926193bf7ce80aab147c3.1424 Call-ID: 686706001c38b1837c7071c84761b300@92.63.138.97 CSeq: 102 OPTIONS Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '686706001c38b1837c7071c84761b300@92.63.138.97' Method: OPTIONS node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INFO sip:92.63.138.97:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK8e6e.a826f774.0 To: sip:92.63.138.97:5060 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-5d60 CSeq: 10 INFO Call-ID: 6ad6496a-7670@92.63.138.100 Content-Length: 0 User-Agent: OpenSIPS (1.5.1-notls (i386/linux)) <-------------> --- (8 headers 0 lines) --- <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK8e6e.a826f774.0;received=92.63.138.100 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-5d60 To: sip:92.63.138.97:5060;tag=as32f5d734 Call-ID: 6ad6496a-7670@92.63.138.100 CSeq: 10 INFO Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INVITE sip:448447041612@92.63.138.97:5060 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKed4a.cdd208.0 Via: SIP/2.0/UDP 87.238.72.153;branch=z9hG4bKed4a.ea2d6ff5.0 Via: SIP/2.0/UDP 87.238.72.155:5060;branch=z9hG4bK1FD0B115F From: "01772814075" ;tag=A5D3EFD4-CA2 To: Date: Tue, 29 Sep 2009 15:50:34 gmt Call-ID: A8BBD8CE-AC4611DE-9806BA8E-F0DAC5A2@87.238.72.155 Supported: timer Min-SE: 1800 User-Agent: MSSGW(B) Allow: INVITE, BYE, CANCEL, ACK CSeq: 101 INVITE Max-Forwards: 13 Remote-Party-ID: ;party=calling;screen=yes;privacy=off Timestamp: 1254239434 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 403 v=0 o=CiscoSystemsSIP-GW-UserAgent 832 5243 IN IP4 87.238.72.155 s=SIP Call c=IN IP4 87.238.72.155 t=0 0 m=audio 16922 RTP/AVP 8 18 4 3 2 0 101 c=IN IP4 87.238.72.155 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:3 GSM/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (23 headers 17 lines) --- == Using SIP RTP CoS mark 5 Sending to 92.63.138.100 : 5060 (no NAT) Using INVITE request as basis request - A8BBD8CE-AC4611DE-9806BA8E-F0DAC5A2@87.238.72.155 No matching peer for '01772814075' from '92.63.138.100:5060' Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 3 Found RTP audio format 2 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 87.238.72.155:16922 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Got unsupported a:fmtp in SDP offer Found audio description format G723 for ID 4 Got unsupported a:fmtp in SDP offer Found audio description format GSM for ID 3 Found audio description format G726-32 for ID 2 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8 (alaw), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 87.238.72.155:16922 Looking for 448447041612 in incoming_magrathea (domain 92.63.138.97) node4*CLI> list_route: hop: list_route: hop: <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKed4a.cdd208.0;received=92.63.138.100 Via: SIP/2.0/UDP 87.238.72.153;branch=z9hG4bKed4a.ea2d6ff5.0 Via: SIP/2.0/UDP 87.238.72.155:5060;branch=z9hG4bK1FD0B115F Record-Route: Record-Route: From: "01772814075" ;tag=A5D3EFD4-CA2 To: Call-ID: A8BBD8CE-AC4611DE-9806BA8E-F0DAC5A2@87.238.72.155 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> node4*CLI> -- Executing AGI("SIP/87.238.72.155-b662cd90", "agi://web0/track.agi?username=emsinternet&campaignName=Website+General&campaignId=204&destination=01925413333&salesTracking=yes&recordCall=yes&analytics=yes") node4*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERPRES()=allowed) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(accountcode)=emsinternet) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(userfield)=204) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERID(name)=Website General) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(amaflags)=billing) node4*CLI> -- AGI Script agi://web0/track.agi?username=emsinternet&campaignName=Website+General&campaignId=204&destination=01925413333&salesTracking=yes&recordCall=yes&analytics=yes completed, returning 0 node4*CLI> -- Executing [448447041612@insight-dialout-external:1] GotoIf("SIP/87.238.72.155-b662cd90", "1?2:5") in new stack node4*CLI> -- Goto (insight-dialout-external,448447041612,2) node4*CLI> -- Executing [448447041612@insight-dialout-external:2] Answer("SIP/87.238.72.155-b662cd90", "") in new stack node4*CLI> Audio is at 92.63.138.97 port 13518 node4*CLI> Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP node4*CLI> <--- Reliably Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKed4a.cdd208.0;received=92.63.138.100 Via: SIP/2.0/UDP 87.238.72.153;branch=z9hG4bKed4a.ea2d6ff5.0 Via: SIP/2.0/UDP 87.238.72.155:5060;branch=z9hG4bK1FD0B115F Record-Route: Record-Route: From: "01772814075" ;tag=A5D3EFD4-CA2 To: ;tag=as7aa5a6d8 Call-ID: A8BBD8CE-AC4611DE-9806BA8E-F0DAC5A2@87.238.72.155 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 265 v=0 o=root 816850128 816850128 IN IP4 92.63.138.97 s=Asterisk PBX 1.6.1.3-rc1 c=IN IP4 92.63.138.97 t=0 0 m=audio 13518 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> ACK sip:448447041612@92.63.138.97:5060 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKed4a.cdd208.2 Via: SIP/2.0/UDP 87.238.72.153;branch=z9hG4bKed4a.ea2d6ff5.2 Via: SIP/2.0/UDP 87.238.72.155:5060;branch=z9hG4bK1FD0B22538 From: ;tag=A5D3EFD4-CA2 To: ;tag=as7aa5a6d8 Date: Tue, 29 Sep 2009 15:50:34 gmt Call-ID: A8BBD8CE-AC4611DE-9806BA8E-F0DAC5A2@87.238.72.155 Max-Forwards: 13 CSeq: 101 ACK Content-Length: 0 <-------------> --- (13 headers 0 lines) --- node4*CLI> -- Executing [448447041612@insight-dialout-external:3] MixMonitor("SIP/87.238.72.155-b662cd90", "voip2-1254239434.651.wav") in new stack node4*CLI> -- Executing [448447041612@insight-dialout-external:4] BackGround("SIP/87.238.72.155-b662cd90", "adinsight-call-recorded") in new stack == Begin MixMonitor Recording SIP/87.238.72.155-b662cd90 node4*CLI> -- Playing 'adinsight-call-recorded.ulaw' (language 'en') node4*CLI> -- Executing [448447041612@insight-dialout-external:5] Dial("SIP/87.238.72.155-b662cd90", "SIP/magrathea-outbound/01925413333,40,CrF(trackSales^448447041612:204:voip2-1254239434.651:01925413333^1)") in new stack node4*CLI> == Using SIP RTP CoS mark 5 node4*CLI> Audio is at 92.63.138.97 port 10684 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP node4*CLI> Reliably Transmitting (no NAT) to 213.166.5.148:5060: INVITE sip:01925413333@sipipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK331a3d99;rport Max-Forwards: 70 From: "Website General" ;tag=as1625127b To: Contact: Call-ID: 4e8abff6187d86403c33edd1579da56d@sipipgw.magrathea.net CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "Website General" ;privacy=off;screen=yes Date: Tue, 29 Sep 2009 15:50:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 265 v=0 o=root 634590609 634590609 IN IP4 92.63.138.97 s=Asterisk PBX 1.6.1.3-rc1 c=IN IP4 92.63.138.97 t=0 0 m=audio 10684 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called magrathea-outbound/01925413333 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK331a3d99;rport=5060 From: "Website General" ;tag=as1625127b To: Call-ID: 4e8abff6187d86403c33edd1579da56d@sipipgw.magrathea.net CSeq: 102 INVITE Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INVITE sip:441223850677@92.63.138.97:5060 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK3728.cc9f53c2.0 Via: SIP/2.0/UDP 213.166.5.129;branch=z9hG4bK3728.d5a824f2.0 Via: SIP/2.0/UDP 213.166.5.133:5060;branch=z9hG4bK579FF442561 From: "01223472000" ;tag=65D99EC8-927 To: Date: Tue, 29 Sep 2009 15:50:38 gmt Call-ID: AB3D39E7-AC4611DE-8483EFB9-710EBE4B@213.166.5.133 Supported: timer,replaces Min-SE: 1800 User-Agent: MSSGW Allow: INVITE, BYE, CANCEL, ACK CSeq: 101 INVITE Max-Forwards: 13 Remote-Party-ID: ;party=calling;screen=yes;privacy=off Timestamp: 1254239438 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 404 v=0 o=CiscoSystemsSIP-GW-UserAgent 9857 5121 IN IP4 213.166.5.133 s=SIP Call c=IN IP4 213.166.5.133 t=0 0 m=audio 19336 RTP/AVP 8 18 4 3 2 0 101 c=IN IP4 213.166.5.133 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:3 GSM/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (23 headers 17 lines) --- == Using SIP RTP CoS mark 5 Sending to 92.63.138.100 : 5060 (no NAT) Using INVITE request as basis request - AB3D39E7-AC4611DE-8483EFB9-710EBE4B@213.166.5.133 No matching peer for '01223472000' from '92.63.138.100:5060' Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 3 Found RTP audio format 2 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 213.166.5.133:19336 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Got unsupported a:fmtp in SDP offer Found audio description format G723 for ID 4 Got unsupported a:fmtp in SDP offer Found audio description format GSM for ID 3 Found audio description format G726-32 for ID 2 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8 (alaw), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 213.166.5.133:19336 Looking for 441223850677 in incoming_magrathea (domain 92.63.138.97) node4*CLI> list_route: hop: list_route: hop: <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK3728.cc9f53c2.0;received=92.63.138.100 Via: SIP/2.0/UDP 213.166.5.129;branch=z9hG4bK3728.d5a824f2.0 Via: SIP/2.0/UDP 213.166.5.133:5060;branch=z9hG4bK579FF442561 Record-Route: Record-Route: From: "01223472000" ;tag=65D99EC8-927 To: Call-ID: AB3D39E7-AC4611DE-8483EFB9-710EBE4B@213.166.5.133 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> node4*CLI> -- Executing AGI("SIP/213.166.5.133-b6635528", "agi://web0/track.agi?username=provisioning&campaignName=Number-1&campaignId=586&destination=08000467086&recordCall=yes") node4*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERPRES()=allowed) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(accountcode)=provisioning) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(userfield)=586) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERID(name)=Number-1) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(amaflags)=billing) node4*CLI> -- AGI Script agi://web0/track.agi?username=provisioning&campaignName=Number-1&campaignId=586&destination=08000467086&recordCall=yes completed, returning 0 node4*CLI> -- Executing [441223850677@insight-dialout-external:1] GotoIf("SIP/213.166.5.133-b6635528", "1?2:5") in new stack node4*CLI> -- Goto (insight-dialout-external,441223850677,2) -- Executing [441223850677@insight-dialout-external:2] Answer("SIP/213.166.5.133-b6635528", "") in new stack node4*CLI> Audio is at 92.63.138.97 port 19216 node4*CLI> Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP node4*CLI> <--- Reliably Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK3728.cc9f53c2.0;received=92.63.138.100 Via: SIP/2.0/UDP 213.166.5.129;branch=z9hG4bK3728.d5a824f2.0 Via: SIP/2.0/UDP 213.166.5.133:5060;branch=z9hG4bK579FF442561 Record-Route: Record-Route: From: "01223472000" ;tag=65D99EC8-927 To: ;tag=as67c27144 Call-ID: AB3D39E7-AC4611DE-8483EFB9-710EBE4B@213.166.5.133 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 265 v=0 o=root 395596501 395596501 IN IP4 92.63.138.97 s=Asterisk PBX 1.6.1.3-rc1 c=IN IP4 92.63.138.97 t=0 0 m=audio 19216 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> ACK sip:441223850677@92.63.138.97:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK3728.cc9f53c2.2 Via: SIP/2.0/UDP 213.166.5.129;branch=z9hG4bK3728.d5a824f2.2 Via: SIP/2.0/UDP 213.166.5.133:5060;branch=z9hG4bK579FF4610CD From: ;tag=65D99EC8-927 To: ;tag=as67c27144 Date: Tue, 29 Sep 2009 15:50:38 gmt Call-ID: AB3D39E7-AC4611DE-8483EFB9-710EBE4B@213.166.5.133 Max-Forwards: 13 CSeq: 101 ACK Content-Length: 0 <-------------> --- (12 headers 0 lines) --- node4*CLI> -- Executing [441223850677@insight-dialout-external:3] MixMonitor("SIP/213.166.5.133-b6635528", "voip2-1254239438.653.wav") in new stack node4*CLI> -- Executing [441223850677@insight-dialout-external:4] BackGround("SIP/213.166.5.133-b6635528", "adinsight-call-recorded") in new stack == Begin MixMonitor Recording SIP/213.166.5.133-b6635528 node4*CLI> -- Playing 'adinsight-call-recorded.ulaw' (language 'en') node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK331a3d99;rport=5060 Record-Route: From: "Website General" ;tag=as1625127b To: ;tag=QD55QpU54QtSK Call-ID: 4e8abff6187d86403c33edd1579da56d@sipipgw.magrathea.net CSeq: 102 INVITE Contact: User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Type: application/sdp Content-Length: 181 v=0 o=- 38575780 0 IN IP4 213.166.5.154 s=Cisco SDP 0 c=IN IP4 213.166.5.154 t=0 0 m=audio 45410 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (13 headers 9 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 213.166.5.154:45410 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 213.166.5.154:45410 node4*CLI> -- SIP/magrathea-outbound-0a23f190 is making progress passing it to SIP/87.238.72.155-b662cd90 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK331a3d99;rport=5060 Record-Route: From: "Website General" ;tag=as1625127b To: ;tag=QD55QpU54QtSK Call-ID: 4e8abff6187d86403c33edd1579da56d@sipipgw.magrathea.net CSeq: 102 INVITE Contact: User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Min-SE: 120 Content-Type: application/sdp Content-Length: 181 v=0 o=- 38575780 0 IN IP4 213.166.5.154 s=Cisco SDP 0 c=IN IP4 213.166.5.154 t=0 0 m=audio 45410 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (14 headers 9 lines) --- list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.148, port 5060 Transmitting (no NAT) to 213.166.5.148:5060: ACK sip:213.166.5.154 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK4cae576b;rport Route: Max-Forwards: 70 From: "Website General" ;tag=as1625127b To: ;tag=QD55QpU54QtSK Contact: Call-ID: 4e8abff6187d86403c33edd1579da56d@sipipgw.magrathea.net CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "Website General" ;privacy=off;screen=yes Content-Length: 0 --- node4*CLI> -- SIP/magrathea-outbound-0a23f190 answered SIP/87.238.72.155-b662cd90 node4*CLI> -- Executing [441223850677@insight-dialout-external:5] Dial("SIP/213.166.5.133-b6635528", "SIP/magrathea-outbound/08000467086,40,Cr") in new stack node4*CLI> == Using SIP RTP CoS mark 5 node4*CLI> Audio is at 92.63.138.97 port 19670 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 213.166.5.148:5060: INVITE sip:08000467086@sipipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK4b8ac4d8;rport Max-Forwards: 70 From: "Number-1" ;tag=as3dc6f2de To: Contact: Call-ID: 1741faff2bd998197351592719e90b9a@sipipgw.magrathea.net CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "Number-1" ;privacy=off;screen=yes Date: Tue, 29 Sep 2009 15:50:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 267 v=0 o=root 1888172409 1888172409 IN IP4 92.63.138.97 s=Asterisk PBX 1.6.1.3-rc1 c=IN IP4 92.63.138.97 t=0 0 m=audio 19670 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- node4*CLI> -- Called magrathea-outbound/08000467086 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK4b8ac4d8;rport=5060 From: "Number-1" ;tag=as3dc6f2de To: Call-ID: 1741faff2bd998197351592719e90b9a@sipipgw.magrathea.net CSeq: 102 INVITE Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK4b8ac4d8;rport=5060 From: "Number-1" ;tag=as3dc6f2de To: ;tag=B0DA21E8-2597 Date: Tue, 29 Sep 2009 15:50:42 gmt Call-ID: 1741faff2bd998197351592719e90b9a@sipipgw.magrathea.net Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Allow-Events: telephone-event Contact: Record-Route: Content-Disposition: session;handling=required Content-Type: application/sdp Content-Length: 250 v=0 o=CiscoSystemsSIP-GW-UserAgent 6809 9175 IN IP4 87.238.72.149 s=SIP Call c=IN IP4 87.238.72.149 t=0 0 m=audio 17854 RTP/AVP 8 101 c=IN IP4 87.238.72.149 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (15 headers 11 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 87.238.72.149:17854 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 87.238.72.149:17854 node4*CLI> -- SIP/magrathea-outbound-b6642160 is making progress passing it to SIP/213.166.5.133-b6635528 node4*CLI> [Sep 29 16:50:47] NOTICE[14773]: rtp.c:1135 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 87.238.72.155 node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INFO sip:92.63.138.97:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK4e6e.95401ae2.0 To: sip:92.63.138.97:5060 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-eb36 CSeq: 10 INFO Call-ID: 6ad6496e-7670@92.63.138.100 Content-Length: 0 User-Agent: OpenSIPS (1.5.1-notls (i386/linux)) <-------------> --- (8 headers 0 lines) --- <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK4e6e.95401ae2.0;received=92.63.138.100 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-eb36 To: sip:92.63.138.97:5060;tag=as4250284d Call-ID: 6ad6496e-7670@92.63.138.100 CSeq: 10 INFO Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK4b8ac4d8;rport=5060 From: "Number-1" ;tag=as3dc6f2de To: ;tag=B0DA21E8-2597 Date: Tue, 29 Sep 2009 15:50:42 gmt Call-ID: 1741faff2bd998197351592719e90b9a@sipipgw.magrathea.net Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Allow-Events: telephone-event Contact: Record-Route: Content-Type: application/sdp Content-Length: 250 v=0 o=CiscoSystemsSIP-GW-UserAgent 6809 9175 IN IP4 87.238.72.149 s=SIP Call c=IN IP4 87.238.72.149 t=0 0 m=audio 17854 RTP/AVP 8 101 c=IN IP4 87.238.72.149 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (14 headers 11 lines) --- list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.148, port 5060 Transmitting (no NAT) to 213.166.5.148:5060: ACK sip:08448000467086@87.238.72.149:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK09c9da61;rport Route: Max-Forwards: 70 From: "Number-1" ;tag=as3dc6f2de To: ;tag=B0DA21E8-2597 Contact: Call-ID: 1741faff2bd998197351592719e90b9a@sipipgw.magrathea.net CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "Number-1" ;privacy=off;screen=yes Content-Length: 0 --- -- SIP/magrathea-outbound-b6642160 answered SIP/213.166.5.133-b6635528 node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> BYE sip:448005244732@92.63.138.97:5060 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK3626.70ac5b34.0 Via: SIP/2.0/UDP 87.238.72.153;branch=z9hG4bK3626.10308f6.0 Via: SIP/2.0/UDP 87.238.72.149:5060;branch=z9hG4bK1694E2EEC8 From: ;tag=B0CFB258-67 To: ;tag=as55576d78 Date: Tue, 29 Sep 2009 15:39:18 gmt Call-ID: 15DB70BE-AC4511DE-B483F3EC-65114225@87.238.72.149 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 13 Timestamp: 1254239460 CSeq: 102 BYE Reason: Q.850;cause=16 Content-Length: 0 <-------------> --- (16 headers 0 lines) --- Sending to 92.63.138.100 : 5060 (no NAT) <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK3626.70ac5b34.0;received=92.63.138.100 Via: SIP/2.0/UDP 87.238.72.153;branch=z9hG4bK3626.10308f6.0 Via: SIP/2.0/UDP 87.238.72.149:5060;branch=z9hG4bK1694E2EEC8 Record-Route: Record-Route: From: ;tag=B0CFB258-67 To: ;tag=as55576d78 Call-ID: 15DB70BE-AC4511DE-B483F3EC-65114225@87.238.72.149 CSeq: 102 BYE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> -- Executing [h@insight-dialout-external:1] Set("SIP/87.238.72.149-b7b7c828", "CDR(outbound)=02082361731") in new stack node4*CLI> Scheduling destruction of SIP dialog '56f09e3810935b215013931351ae50b7@sipipgw.magrathea.net' in 6400 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.148, port 5060 Reliably Transmitting (no NAT) to 213.166.5.148:5060: BYE sip:213.166.5.153 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK41bb7b05;rport Route: Max-Forwards: 70 From: "Internet" ;tag=as750abc61 To: ;tag=0F97teK5jXa8B Call-ID: 56f09e3810935b215013931351ae50b7@sipipgw.magrathea.net CSeq: 103 BYE User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "Internet" ;privacy=off;screen=yes X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- == Spawn extension (insight-dialout-external, 448005244732, 5) exited non-zero on 'SIP/87.238.72.149-b7b7c828' node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK41bb7b05;rport=5060 From: "Internet" ;tag=as750abc61 To: ;tag=0F97teK5jXa8B Call-ID: 56f09e3810935b215013931351ae50b7@sipipgw.magrathea.net CSeq: 103 BYE User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '56f09e3810935b215013931351ae50b7@sipipgw.magrathea.net' Method: INVITE Really destroying SIP dialog '15DB70BE-AC4511DE-B483F3EC-65114225@87.238.72.149' Method: BYE node4*CLI> Reliably Transmitting (no NAT) to 213.166.5.148:5060: OPTIONS sip:sipipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK2d0e5a78;rport Max-Forwards: 70 From: "asterisk" ;tag=as592ca873 To: Contact: Call-ID: 796a3dad105c0cd1581717f7740bb322@92.63.138.97 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.1.3-rc1 Date: Tue, 29 Sep 2009 15:51:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 403 OPTIONS not supported Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK2d0e5a78;rport=5060 From: "asterisk" ;tag=as592ca873 To: ;tag=9a264c9a00f926193bf7ce80aab147c3.7c6a Call-ID: 796a3dad105c0cd1581717f7740bb322@92.63.138.97 CSeq: 102 OPTIONS Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '796a3dad105c0cd1581717f7740bb322@92.63.138.97' Method: OPTIONS node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> BYE sip:02088462000@92.63.138.97 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 213.166.5.148;branch=z9hG4bK0ea1.2cb506e1.0 Via: SIP/2.0/UDP 87.238.72.133:5070;branch=z9hG4bK0ea1.7dce6406.0 Via: SIP/2.0/UDP 87.238.72.134;rport=5060;branch=z9hG4bK2g3310r95rKyS Max-Forwards: 68 From: ;tag=eK08Dm90p4cmB To: "Number-5" ;tag=as7d9d02d7 Call-ID: 07f24a2a3502e10b0335f3e3628d310e@sipipgw.magrathea.net CSeq: 121001403 BYE User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Sending to 213.166.5.148 : 5060 (no NAT) <--- Transmitting (no NAT) to 213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 213.166.5.148;branch=z9hG4bK0ea1.2cb506e1.0;received=213.166.5.148 Via: SIP/2.0/UDP 87.238.72.133:5070;branch=z9hG4bK0ea1.7dce6406.0 V node4*CLI> ia: SIP/2.0/UDP 87.238.72.134;rport=5060;branch=z9hG4bK2g3310r95rKyS Record-Route: From: ;tag=eK08Dm90p4cmB To: "Number-5" ;tag=as7d9d02d7 Call-ID: 07f24a2a3502e10b0335f3e3628d310e@sipipgw.magrathea.net CSeq: 121001403 BYE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> -- Executing [h@insight-dialout-external:1] Set("SIP/213.166.5.140-b660f528", "CDR(outbound)=02077360193") in new stack node4*CLI> == Spawn extension (insight-dialout-external, 442033936468, 5) exited non-zero on 'SIP/213.166.5.140-b660f528' Scheduling destruction of SIP dialog '97D5636A-AC4611DE-A8D9D31F-EFD16753@213.166.5.140' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 92.63.138.100, port 5060 Reliably Transmitting (no NAT) to 92.63.138.100:5060: BYE sip:02088462000@213.166.5.140:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK63132908;rport Route: , Max-Forwards: 70 From: ;tag=as2398868e To: "02088462000" ;tag=C67080C4-206A Call-ID: 97D5636A-AC4611DE-A8D9D31F-EFD16753@213.166.5.140 CSeq: 102 BYE User-Agent: Asterisk PBX 1.6.1.3-rc1 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- node4*CLI> == MixMonitor close filestream node4*CLI> == End MixMonitor Recording SIP/213.166.5.140-b660f528 node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK63132908;rport=5060 From: ;tag=as2398868e To: ;tag=C67080C4-206A Date: Tue, 29 Sep 2009 15:51:20 gmt Call-ID: 97D5636A-AC4611DE-A8D9D31F-EFD16753@213.166.5.140 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 BYE Reason: Q.850;cause=16 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '97D5636A-AC4611DE-A8D9D31F-EFD16753@213.166.5.140' Method: ACK Really destroying SIP dialog '07f24a2a3502e10b0335f3e3628d310e@sipipgw.magrathea.net' Method: BYE node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INFO sip:92.63.138.97:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKf08e.cd9213f4.0 To: sip:92.63.138.97:5060 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-933a CSeq: 10 INFO Call-ID: 6ad64972-7670@92.63.138.100 Content-Length: 0 User-Agent: OpenSIPS (1.5.1-notls (i386/linux)) <-------------> --- (8 headers 0 lines) --- <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKf08e.cd9213f4.0;received=92.63.138.100 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-933a To: sip:92.63.138.97:5060;tag=as2a7b02f0 Call-ID: 6ad64972-7670@92.63.138.100 CSeq: 10 INFO Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> BYE sip:442034118274@92.63.138.97:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK710a.dba3a5e2.0 Via: SIP/2.0/UDP 213.166.5.129;branch=z9hG4bK710a.986b4415.0 Via: SIP/2.0/UDP 213.166.5.133:5060;branch=z9hG4bK57A004F186D From: ;tag=65D81A60-F23 To: ;tag=as500f55d1 Date: Tue, 29 Sep 2009 15:48:59 gmt Call-ID: 6FF853C0-AC4611DE-BF66EFB9-710EBE4B@213.166.5.133 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 13 Timestamp: 1254239484 CSeq: 102 BYE Reason: Q.850;cause=16 Content-Length: 0 <-------------> --- (15 headers 0 lines) --- Sending to 92.63.138.100 : 5060 (no NAT) <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK710a.dba3a5e2.0;received=92.63.138.100 Via: SIP/2.0/UDP 213.166.5.129;branch=z9hG4bK710a.986b4415.0 Via: SIP/2.0/UDP 213.166.5.133:5060;branch=z9hG4bK57A004F186D Record-Route: From: ;tag=65D81A60-F23 To: ;tag=as500f55d1 Call-ID: 6FF853C0-AC4611DE-BF66EFB9-710EBE4B@213.166.5.133 CSeq: 102 BYE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> -- Executing [h@insight-dialout-external:1] Set("SIP/213.166.5.133-b662b550", "CDR(outbound)=02072266066") in new stack node4*CLI> Scheduling destruction of SIP dialog '76b7bf8824076b182f31c06f02ee4ff7@sipipgw.magrathea.net' in 6400 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.148, port 5060 Reliably Transmitting (no NAT) to 213.166.5.148:5060: BYE sip:213.166.5.153 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK675e3ee4;rport Route: Max-Forwards: 70 From: "Number-2" ;tag=as5a444872 To: ;tag=437gjpQmQjr7D Call-ID: 76b7bf8824076b182f31c06f02ee4ff7@sipipgw.magrathea.net CSeq: 103 BYE User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "Number-2" ;privacy=off;screen=yes X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- == Spawn extension (insight-dialout-external, 442034118274, 5) exited non-zero on 'SIP/213.166.5.133-b662b550' node4*CLI> == MixMonitor close filestream node4*CLI> == End MixMonitor Recording SIP/213.166.5.133-b662b550 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK675e3ee4;rport=5060 From: "Number-2" ;tag=as5a444872 To: ;tag=437gjpQmQjr7D Call-ID: 76b7bf8824076b182f31c06f02ee4ff7@sipipgw.magrathea.net CSeq: 103 BYE User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '76b7bf8824076b182f31c06f02ee4ff7@sipipgw.magrathea.net' Method: INVITE Really destroying SIP dialog '6FF853C0-AC4611DE-BF66EFB9-710EBE4B@213.166.5.133' Method: BYE node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INFO sip:92.63.138.97:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKb08e.d4ddfa97.0 To: sip:92.63.138.97:5060 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-256c CSeq: 10 INFO Call-ID: 6ad64976-7670@92.63.138.100 Content-Length: 0 User-Agent: OpenSIPS (1.5.1-notls (i386/linux)) <-------------> --- (8 headers 0 lines) --- <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKb08e.d4ddfa97.0;received=92.63.138.100 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-256c To: sip:92.63.138.97:5060;tag=as67198792 Call-ID: 6ad64976-7670@92.63.138.100 CSeq: 10 INFO Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> Reliably Transmitting (no NAT) to 213.166.5.148:5060: OPTIONS sip:sipipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK68d65ac3;rport Max-Forwards: 70 From: "asterisk" ;tag=as32837a35 To: Contact: Call-ID: 7d271183603153ca10247729784e6df6@92.63.138.97 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.1.3-rc1 Date: Tue, 29 Sep 2009 15:52:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 403 OPTIONS not supported Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK68d65ac3;rport=5060 From: "asterisk" ;tag=as32837a35 To: ;tag=9a264c9a00f926193bf7ce80aab147c3.b614 Call-ID: 7d271183603153ca10247729784e6df6@92.63.138.97 CSeq: 102 OPTIONS Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '7d271183603153ca10247729784e6df6@92.63.138.97' Method: OPTIONS node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INFO sip:92.63.138.97:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK8c7e.10940c67.0 To: sip:92.63.138.97:5060 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-9b13 CSeq: 10 INFO Call-ID: 6ad6497a-7670@92.63.138.100 Content-Length: 0 User-Agent: OpenSIPS (1.5.1-notls (i386/linux)) <-------------> --- (8 headers 0 lines) --- <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK8c7e.10940c67.0;received=92.63.138.100 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-9b13 To: sip:92.63.138.97:5060;tag=as47aa5a25 Call-ID: 6ad6497a-7670@92.63.138.100 CSeq: 10 INFO Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INVITE sip:442033931863@92.63.138.97:5060 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK20c5.2afe0866.0 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bK20c5.3e88a631.0 Via: SIP/2.0/UDP 213.166.5.140:5060;branch=z9hG4bK50466C4DE Remote-Party-ID: ;party=calling;screen=yes;privacy=off From: "07540693656" ;tag=C672D7AC-1238 To: Date: Tue, 29 Sep 2009 15:52:39 gmt Call-ID: F33839A5-AC4611DE-AE77D31F-EFD16753@213.166.5.140 Supported: timer,resource-priority,replaces Min-SE: 1800 User-Agent: MSSGW Allow: INVITE, BYE, CANCEL, ACK CSeq: 101 INVITE Max-Forwards: 13 Timestamp: 1254239559 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 418 v=0 o=CiscoSystemsSIP-GW-UserAgent 2272 8344 IN IP4 213.166.5.140 s=SIP Call c=IN IP4 213.166.5.140 t=0 0 m=audio 19166 RTP/AVP 8 18 4 3 98 0 101 c=IN IP4 213.166.5.140 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:4 G723/8000 a=fmtp:4 bitrate=6.3;annexa=no a=rtpmap:3 GSM/8000 a=rtpmap:98 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (24 headers 17 lines) --- == Using SIP RTP CoS mark 5 Sending to 92.63.138.100 : 5060 (no NAT) Using INVITE request as basis request - F33839A5-AC4611DE-AE77D31F-EFD16753@213.166.5.140 No matching peer for '07540693656' from '92.63.138.100:5060' Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 3 Found RTP audio format 98 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 213.166.5.140:19166 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Got unsupported a:fmtp in SDP offer Found audio description format G723 for ID 4 Got unsupported a:fmtp in SDP offer Found audio description format GSM for ID 3 Found audio description format G726-32 for ID 98 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8 (alaw), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 213.166.5.140:19166 Looking for 442033931863 in incoming_magrathea (domain 92.63.138.97) node4*CLI> list_route: hop: list_route: hop: <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK20c5.2afe0866.0;received=92.63.138.100 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bK20c5.3e88a631.0 Via: SIP/2.0/UDP 213.166.5.140:5060;branch=z9hG4bK50466C4DE Record-Route: Record-Route: From: "07540693656" ;tag=C672D7AC-1238 To: Call-ID: F33839A5-AC4611DE-AE77D31F-EFD16753@213.166.5.140 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> node4*CLI> -- Executing AGI("SIP/213.166.5.140-b6608498", "agi://web0/track.agi?username=provisioning&campaignName=Number-1&campaignId=490&destination=02070934488&recordCall=yes") node4*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERPRES()=allowed) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(accountcode)=provisioning) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(userfield)=490) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERID(name)=Number-1) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(amaflags)=billing) node4*CLI> -- AGI Script agi://web0/track.agi?username=provisioning&campaignName=Number-1&campaignId=490&destination=02070934488&recordCall=yes completed, returning 0 node4*CLI> -- Executing [442033931863@insight-dialout-external:1] GotoIf("SIP/213.166.5.140-b6608498", "1?2:5") in new stack -- Goto (insight-dialout-external,442033931863,2) -- Executing [442033931863@insight-dialout-external:2] Answer("SIP/213.166.5.140-b6608498", "") in new stack node4*CLI> Audio is at 92.63.138.97 port 19756 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP node4*CLI> <--- Reliably Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK20c5.2afe0866.0;received=92.63.138.100 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bK20c5.3e88a631.0 Via: SIP/2.0/UDP 213.166.5.140:5060;branch=z9hG4bK50466C4DE Record-Route: Record-Route: From: "07540693656" ;tag=C672D7AC-1238 To: ;tag=as7079980e Call-ID: F33839A5-AC4611DE-AE77D31F-EFD16753@213.166.5.140 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 267 v=0 o=root 1436922626 1436922626 IN IP4 92.63.138.97 s=Asterisk PBX 1.6.1.3-rc1 c=IN IP4 92.63.138.97 t=0 0 m=audio 19756 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> ACK sip:442033931863@92.63.138.97:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK20c5.2afe0866.2 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bK20c5.3e88a631.2 Via: SIP/2.0/UDP 213.166.5.140:5060;branch=z9hG4bK50466D1F7D From: ;tag=C672D7AC-1238 To: ;tag=as7079980e Date: Tue, 29 Sep 2009 15:52:39 gmt Call-ID: F33839A5-AC4611DE-AE77D31F-EFD16753@213.166.5.140 Max-Forwards: 13 CSeq: 101 ACK Allow-Events: telephone-event Content-Length: 0 <-------------> --- (13 headers 0 lines) --- node4*CLI> -- Executing [442033931863@insight-dialout-external:3] MixMonitor("SIP/213.166.5.140-b6608498", "voip2-1254239559.655.wav") in new stack node4*CLI> == Begin MixMonitor Recording SIP/213.166.5.140-b6608498 -- Executing [442033931863@insight-dialout-external:4] BackGround("SIP/213.166.5.140-b6608498", "adinsight-call-recorded") in new stack node4*CLI> -- Playing 'adinsight-call-recorded.ulaw' (language 'en') node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> BYE sip:441223850677@92.63.138.97:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK0728.27d5a4f1.0 Via: SIP/2.0/UDP 213.166.5.129;branch=z9hG4bK0728.f4e9566.0 Via: SIP/2.0/UDP 213.166.5.133:5060;branch=z9hG4bK57A026ED5B From: ;tag=65D99EC8-927 To: ;tag=as67c27144 Date: Tue, 29 Sep 2009 15:50:38 gmt Call-ID: AB3D39E7-AC4611DE-8483EFB9-710EBE4B@213.166.5.133 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 13 Timestamp: 1254239562 CSeq: 102 BYE Reason: Q.850;cause=16 Content-Length: 0 <-------------> --- (15 headers 0 lines) --- Sending to 92.63.138.100 : 5060 (no NAT) <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK0728.27d5a4f1.0;received=92.63.138.100 Via: SIP/2.0/UDP 213.166.5.129;branch=z9hG4bK0728.f4e9566.0 Via: SIP/2.0/UDP 213.166.5.133:5060;branch=z9hG4bK57A026ED5B Record-Route: From: ;tag=65D99EC8-927 To: ;tag=as67c27144 Call-ID: AB3D39E7-AC4611DE-8483EFB9-710EBE4B@213.166.5.133 CSeq: 102 BYE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> -- Executing [h@insight-dialout-external:1] Set("SIP/213.166.5.133-b6635528", "CDR(outbound)=08000467086") in new stack node4*CLI> Scheduling destruction of SIP dialog '1741faff2bd998197351592719e90b9a@sipipgw.magrathea.net' in 6400 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.148, port 5060 Reliably Transmitting (no NAT) to 213.166.5.148:5060: BYE sip:08448000467086@87.238.72.149:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK631cc5d8;rport Route: Max-Forwards: 70 From: "Number-1" ;tag=as3dc6f2de To: ;tag=B0DA21E8-2597 Call-ID: 1741faff2bd998197351592719e90b9a@sipipgw.magrathea.net CSeq: 103 BYE User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "Number-1" ;privacy=off;screen=yes X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- == Spawn extension (insight-dialout-external, 441223850677, 5) exited non-zero on 'SIP/213.166.5.133-b6635528' node4*CLI> == MixMonitor close filestream == End MixMonitor Recording SIP/213.166.5.133-b6635528 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK631cc5d8;rport=5060 From: "Number-1" ;tag=as3dc6f2de To: ;tag=B0DA21E8-2597 Date: Tue, 29 Sep 2009 15:52:42 gmt Call-ID: 1741faff2bd998197351592719e90b9a@sipipgw.magrathea.net Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 103 BYE <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '1741faff2bd998197351592719e90b9a@sipipgw.magrathea.net' Method: INVITE Really destroying SIP dialog 'AB3D39E7-AC4611DE-8483EFB9-710EBE4B@213.166.5.133' Method: BYE node4*CLI> -- Executing [442033931863@insight-dialout-external:5] Dial("SIP/213.166.5.140-b6608498", "SIP/magrathea-outbound/02070934488,40,Cr") in new stack node4*CLI> == Using SIP RTP CoS mark 5 node4*CLI> Audio is at 92.63.138.97 port 11762 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 213.166.5.148:5060: INVITE sip:02070934488@sipipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK65e2eebf;rport Max-Forwards: 70 From: "Number-1" ;tag=as06213673 To: Contact: Call-ID: 499ec2950521784b11c6325a1c1bacfb@sipipgw.magrathea.net CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "Number-1" ;privacy=off;screen=yes Date: Tue, 29 Sep 2009 15:52:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 267 v=0 o=root 1877185412 1877185412 IN IP4 92.63.138.97 s=Asterisk PBX 1.6.1.3-rc1 c=IN IP4 92.63.138.97 t=0 0 m=audio 11762 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called magrathea-outbound/02070934488 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK65e2eebf;rport=5060 From: "Number-1" ;tag=as06213673 To: Call-ID: 499ec2950521784b11c6325a1c1bacfb@sipipgw.magrathea.net CSeq: 102 INVITE Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK65e2eebf;rport=5060 Record-Route: From: "Number-1" ;tag=as06213673 To: ;tag=UUrgDXrX3jQ2g Call-ID: 499ec2950521784b11c6325a1c1bacfb@sipipgw.magrathea.net CSeq: 102 INVITE Contact: User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Type: application/sdp Content-Length: 181 v=0 o=- 53378361 0 IN IP4 213.166.5.153 s=Cisco SDP 0 c=IN IP4 213.166.5.153 t=0 0 m=audio 39694 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (13 headers 9 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 213.166.5.153:39694 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 213.166.5.153:39694 node4*CLI> -- SIP/magrathea-outbound-b662a1f8 is making progress passing it to SIP/213.166.5.140-b6608498 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK65e2eebf;rport=5060 Record-Route: From: "Number-1" ;tag=as06213673 To: ;tag=UUrgDXrX3jQ2g Call-ID: 499ec2950521784b11c6325a1c1bacfb@sipipgw.magrathea.net CSeq: 102 INVITE Contact: User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Min-SE: 120 Content-Type: application/sdp Content-Length: 181 v=0 o=- 53378361 0 IN IP4 213.166.5.153 s=Cisco SDP 0 c=IN IP4 213.166.5.153 t=0 0 m=audio 39694 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (14 headers 9 lines) --- list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.148, port 5060 Transmitting (no NAT) to 213.166.5.148:5060: ACK sip:213.166.5.153 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK35d72197;rport Route: Max-Forwards: 70 From: "Number-1" ;tag=as06213673 To: ;tag=UUrgDXrX3jQ2g Contact: Call-ID: 499ec2950521784b11c6325a1c1bacfb@sipipgw.magrathea.net CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "Number-1" ;privacy=off;screen=yes Content-Length: 0 --- node4*CLI> -- SIP/magrathea-outbound-b662a1f8 answered SIP/213.166.5.140-b6608498 node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INFO sip:92.63.138.97:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK4c7e.8ab5a0f.0 To: sip:92.63.138.97:5060 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-2d45 CSeq: 10 INFO Call-ID: 6ad6497e-7670@92.63.138.100 Content-Length: 0 User-Agent: OpenSIPS (1.5.1-notls (i386/linux)) <-------------> --- (8 headers 0 lines) --- <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK4c7e.8ab5a0f.0;received=92.63.138.100 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-2d45 To: sip:92.63.138.97:5060;tag=as1021b30f Call-ID: 6ad6497e-7670@92.63.138.100 CSeq: 10 INFO Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> BYE sip:01772814075@92.63.138.97 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 213.166.5.148;branch=z9hG4bK5fd1.d846d211.0 Via: SIP/2.0/UDP 213.166.5.139:5070;branch=z9hG4bK5fd1.9eb17505.0 Via: SIP/2.0/UDP 213.166.5.154;rport=5060;branch=z9hG4bKy7vHmayQt8Dtr Max-Forwards: 68 From: ;tag=QD55QpU54QtSK To: "Website General" ;tag=as1625127b Call-ID: 4e8abff6187d86403c33edd1579da56d@sipipgw.magrathea.net CSeq: 121001462 BYE User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Sending to 213.166.5.148 : 5060 (no NAT) <--- Transmitting (no NAT) to 213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 213.166.5.148;branch=z9hG4bK5fd1.d846d211.0;received=213.166.5.148 Via: SIP/2.0/UDP 213.166.5.139:5070;branch=z9hG4bK5fd1.9eb17505.0 Via: SIP/2.0/UDP 213.166.5.154;rport=5060;branch=z9hG4bKy7vHmayQt8Dtr Record-Route: From: ;tag=QD55QpU54QtSK To: "Website General" ;tag=as1625127b Call-ID: 4e8abff6187d86403c33edd1579da56d@sipipgw.magrathea.net CSeq: 121001462 BYE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> -- Executing [h@insight-dialout-external:1] Set("SIP/87.238.72.155-b662cd90", "CDR(outbound)=01925413333") in new stack node4*CLI> == Spawn extension (insight-dialout-external, 448447041612, 5) exited non-zero on 'SIP/87.238.72.155-b662cd90' Scheduling destruction of SIP dialog 'A8BBD8CE-AC4611DE-9806BA8E-F0DAC5A2@87.238.72.155' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 92.63.138.100, port 5060 Reliably Transmitting (no NAT) to 92.63.138.100:5060: BYE sip:01772814075@87.238.72.155:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK2de755c9;rport Route: , Max-Forwards: 70 From: ;tag=as7aa5a6d8 To: "01772814075" ;tag=A5D3EFD4-CA2 Call-ID: A8BBD8CE-AC4611DE-9806BA8E-F0DAC5A2@87.238.72.155 CSeq: 102 BYE User-Agent: Asterisk PBX 1.6.1.3-rc1 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- node4*CLI> == MixMonitor close filestream node4*CLI> == End MixMonitor Recording SIP/87.238.72.155-b662cd90 node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK2de755c9;rport=5060 From: ;tag=as7aa5a6d8 To: ;tag=A5D3EFD4-CA2 Date: Tue, 29 Sep 2009 15:53:18 gmt Call-ID: A8BBD8CE-AC4611DE-9806BA8E-F0DAC5A2@87.238.72.155 Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 102 BYE <-------------> --- (9 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '4e8abff6187d86403c33edd1579da56d@sipipgw.magrathea.net' Method: BYE Really destroying SIP dialog 'A8BBD8CE-AC4611DE-9806BA8E-F0DAC5A2@87.238.72.155' Method: ACK node4*CLI> Reliably Transmitting (no NAT) to 213.166.5.148:5060: OPTIONS sip:sipipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK48660982;rport Max-Forwards: 70 From: "asterisk" ;tag=as35bf5183 To: Contact: Call-ID: 26bbfa3d51b2c95372e6026d48303b22@92.63.138.97 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.1.3-rc1 Date: Tue, 29 Sep 2009 15:53:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 403 OPTIONS not supported Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK48660982;rport=5060 From: "asterisk" ;tag=as35bf5183 To: ;tag=9a264c9a00f926193bf7ce80aab147c3.39f0 Call-ID: 26bbfa3d51b2c95372e6026d48303b22@92.63.138.97 CSeq: 102 OPTIONS Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '26bbfa3d51b2c95372e6026d48303b22@92.63.138.97' Method: OPTIONS node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INFO sip:92.63.138.97:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK175f.85c61782.0 To: sip:92.63.138.97:5060 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-c449 CSeq: 10 INFO Call-ID: 6ad64982-7670@92.63.138.100 Content-Length: 0 User-Agent: OpenSIPS (1.5.1-notls (i386/linux)) <-------------> --- (8 headers 0 lines) --- <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK175f.85c61782.0;received=92.63.138.100 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-c449 To: sip:92.63.138.97:5060;tag=as584ccef5 Call-ID: 6ad64982-7670@92.63.138.100 CSeq: 10 INFO Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> BYE sip:01619734347@92.63.138.97 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 213.166.5.148;branch=z9hG4bK5a87.44da7056.0 Via: SIP/2.0/UDP 213.166.5.139:5070;branch=z9hG4bK5a87.f2ab7ed6.0 Via: SIP/2.0/UDP 213.166.5.153;rport=5060;branch=z9hG4bK0HHv7v450FNac Max-Forwards: 68 From: ;tag=ZHXZ6yX94QS3c To: "direct traffic" ;tag=as5d938583 Call-ID: 19d637887454386e0f55e269610cd8ad@sipipgw.magrathea.net CSeq: 121001464 BYE User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Sending to 213.166.5.148 : 5060 (no NAT) <--- Transmitting (no NAT) to 213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 213.166.5.148;branch=z9hG4bK5a87.44da7056.0;received=213.166.5.148 Via: SIP/2.0/UDP 213.166.5.139:5070;branch=z9hG4bK5a87.f2ab7ed6.0 Via: SIP/2.0/UDP 213.166.5.153;rport=5060;branch=z9hG4bK0HHv7v450FNac Record-Route: From: ;tag=ZHXZ6yX94QS3c To: "direct traffic" ;tag=as5d938583 Call-ID: 19d637887454386e0f55e269610cd8ad@sipipgw.magrathea.net CSeq: 121001464 BYE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> -- Executing [h@insight-dialout-external:1] Set("SIP/87.238.72.149-b664a630", "CDR(outbound)=02089095222") in new stack node4*CLI> == Spawn extension (insight-dialout-external, 448005244040, 5) exited non-zero on 'SIP/87.238.72.149-b664a630' Scheduling destruction of SIP dialog '92398592-AC4611DE-BF0EF3EC-65114225@87.238.72.149' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 92.63.138.100, port 5060 Reliably Transmitting (no NAT) to 92.63.138.100:5060: BYE sip:01619734347@87.238.72.149:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK0652d007;rport Route: , Max-Forwards: 70 From: ;tag=as2d71f2b0 To: "01619734347" ;tag=B0D96F1C-159D Call-ID: 92398592-AC4611DE-BF0EF3EC-65114225@87.238.72.149 CSeq: 102 BYE User-Agent: Asterisk PBX 1.6.1.3-rc1 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- node4*CLI> == MixMonitor close filestream node4*CLI> == End MixMonitor Recording SIP/87.238.72.149-b664a630 node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK0652d007;rport=5060 From: ;tag=as2d71f2b0 To: ;tag=B0D96F1C-159D Date: Tue, 29 Sep 2009 15:53:22 gmt Call-ID: 92398592-AC4611DE-BF0EF3EC-65114225@87.238.72.149 Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 102 BYE <-------------> --- (9 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '92398592-AC4611DE-BF0EF3EC-65114225@87.238.72.149' Method: ACK Really destroying SIP dialog '19d637887454386e0f55e269610cd8ad@sipipgw.magrathea.net' Method: BYE node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INVITE sip:448081201310@92.63.138.97:5060 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKee4d.cff31a21.0 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bKee4d.f5360672.0 Via: SIP/2.0/UDP 213.166.5.140:5060;branch=z9hG4bK5047032372 Remote-Party-ID: ;party=calling;screen=yes;privacy=off From: "02077311072" ;tag=C673CEF8-12CE To: Date: Tue, 29 Sep 2009 15:53:42 gmt Call-ID: 18F448DF-AC4711DE-B0B1D31F-EFD16753@213.166.5.140 Supported: timer,resource-priority,replaces Min-SE: 1800 User-Agent: MSSGW Allow: INVITE, BYE, CANCEL, ACK CSeq: 101 INVITE Max-Forwards: 13 Timestamp: 1254239622 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 418 v=0 o=CiscoSystemsSIP-GW-UserAgent 4254 4831 IN IP4 213.166.5.140 s=SIP Call c=IN IP4 213.166.5.140 t=0 0 m=audio 18198 RTP/AVP 8 18 4 3 98 0 101 c=IN IP4 213.166.5.140 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:4 G723/8000 a=fmtp:4 bitrate=6.3;annexa=no a=rtpmap:3 GSM/8000 a=rtpmap:98 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (24 headers 17 lines) --- node4*CLI> == Using SIP RTP CoS mark 5 Sending to 92.63.138.100 : 5060 (no NAT) Using INVITE request as basis request - 18F448DF-AC4711DE-B0B1D31F-EFD16753@213.166.5.140 No matching peer for '02077311072' from '92.63.138.100:5060' Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 3 Found RTP audio format 98 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 213.166.5.140:18198 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Got unsupported a:fmtp in SDP offer Found audio description format G723 for ID 4 Got unsupported a:fmtp in SDP offer Found audio description format GSM for ID 3 Found audio description format G726-32 for ID 98 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8 (alaw), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 213.166.5.140:18198 Looking for 448081201310 in incoming_magrathea (domain 92.63.138.97) node4*CLI> list_route: hop: list_route: hop: node4*CLI> <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKee4d.cff31a21.0;received=92.63.138.100 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bKee4d.f5360672.0 Via: SIP/2.0/UDP 213.166.5.140:5060;branch=z9hG4bK5047032372 Record-Route: Record-Route: From: "02077311072" ;tag=C673CEF8-12CE To: node4*CLI> Call-ID: 18F448DF-AC4711DE-B0B1D31F-EFD16753@213.166.5.140 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> node4*CLI> -- Executing AGI("SIP/213.166.5.140-b6635528", "agi://web0/track.agi?username=provisioning&campaignName=C9486-drainageex-london&campaignId=375&destination=01425475777&recordCall=yes") node4*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERPRES()=allowed) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(accountcode)=provisioning) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(userfield)=375) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERID(name)=C9486-drainageex-london) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(amaflags)=billing) node4*CLI> -- AGI Script agi://web0/track.agi?username=provisioning&campaignName=C9486-drainageex-london&campaignId=375&destination=01425475777&recordCall=yes completed, returning 0 node4*CLI> -- Executing [448081201310@insight-dialout-external:1] GotoIf("SIP/213.166.5.140-b6635528", "1?2:5") in new stack -- Goto (insight-dialout-external,448081201310,2) -- Executing [448081201310@insight-dialout-external:2] Answer("SIP/213.166.5.140-b6635528", "") in new stack node4*CLI> Audio is at 92.63.138.97 port 15202 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP node4*CLI> <--- Reliably Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKee4d.cff31a21.0;received=92.63.138.100 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bKee4d.f5360672.0 Via: SIP/2.0/UDP 213.166.5.140:5060;branch=z9hG4bK5047032372 Record-Route: Record-Route: From: "02077311072" ;tag=C673CEF8-12CE To: ;tag=as16c295b7 Call-ID: 18F448DF-AC4711DE-B0B1D31F-EFD16753@213.166.5.140 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 267 v=0 o=root 1083260246 1083260246 IN IP4 92.63.138.97 s=Asterisk PBX 1.6.1.3-rc1 c=IN IP4 92.63.138.97 t=0 0 m=audio 15202 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> ACK sip:448081201310@92.63.138.97:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKee4d.cff31a21.2 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bKee4d.f5360672.2 Via: SIP/2.0/UDP 213.166.5.140:5060;branch=z9hG4bK5047041230 From: ;tag=C673CEF8-12CE To: ;tag=as16c295b7 Date: Tue, 29 Sep 2009 15:53:42 gmt Call-ID: 18F448DF-AC4711DE-B0B1D31F-EFD16753@213.166.5.140 Max-Forwards: 13 CSeq: 101 ACK Allow-Events: telephone-event Content-Length: 0 <-------------> --- (13 headers 0 lines) --- node4*CLI> -- Executing [448081201310@insight-dialout-external:3] MixMonitor("SIP/213.166.5.140-b6635528", "voip2-1254239622.657.wav") in new stack node4*CLI> -- Executing [448081201310@insight-dialout-external:4] BackGround("SIP/213.166.5.140-b6635528", "adinsight-call-recorded") in new stack == Begin MixMonitor Recording SIP/213.166.5.140-b6635528 node4*CLI> -- Playing 'adinsight-call-recorded.ulaw' (language 'en') node4*CLI> -- Executing [448081201310@insight-dialout-external:5] Dial("SIP/213.166.5.140-b6635528", "SIP/magrathea-outbound/01425475777,40,Cr") in new stack node4*CLI> == Using SIP RTP CoS mark 5 node4*CLI> Audio is at 92.63.138.97 port 17128 node4*CLI> Adding codec 0x8 (alaw) to SDP node4*CLI> Adding non-codec 0x1 (telephone-event) to SDP node4*CLI> Reliably Transmitting (no NAT) to 213.166.5.148:5060: INVITE sip:01425475777@sipipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK71cc1a2d;rport Max-Forwards: 70 From: "C9486-drainageex-london" ;tag=as7f4ad4f6 To: Contact: Call-ID: 01faa6fa6a0303774a75319231829daa@sipipgw.magrathea.net CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "C9486-drainageex-london" ;privacy=off;screen=yes Date: Tue, 29 Sep 2009 15:53:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 265 v=0 o=root 849028161 849028161 IN IP4 92.63.138.97 s=Asterisk PBX 1.6.1.3-rc1 c=IN IP4 92.63.138.97 t=0 0 m=audio 17128 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- node4*CLI> -- Called magrathea-outbound/01425475777 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK71cc1a2d;rport=5060 From: "C9486-drainageex-london" ;tag=as7f4ad4f6 To: Call-ID: 01faa6fa6a0303774a75319231829daa@sipipgw.magrathea.net CSeq: 102 INVITE Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK71cc1a2d;rport=5060 Record-Route: From: "C9486-drainageex-london" ;tag=as7f4ad4f6 To: ;tag=S9672ZrHt475F Call-ID: 01faa6fa6a0303774a75319231829daa@sipipgw.magrathea.net CSeq: 102 INVITE Contact: User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Type: application/sdp Content-Length: 181 v=0 o=- 38578177 0 IN IP4 213.166.5.153 s=Cisco SDP 0 c=IN IP4 213.166.5.153 t=0 0 m=audio 41138 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (13 headers 9 lines) --- node4*CLI> Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 213.166.5.153:41138 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 213.166.5.153:41138 -- SIP/magrathea-outbound-0a1bd1f8 is making progress passing it to SIP/213.166.5.140-b6635528 node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INFO sip:92.63.138.97:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKd65f.6eecb976.0 To: sip:92.63.138.97:5060 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-721f CSeq: 10 INFO Call-ID: 6ad64986-7670@92.63.138.100 Content-Length: 0 User-Agent: OpenSIPS (1.5.1-notls (i386/linux)) <-------------> --- (8 headers 0 lines) --- node4*CLI> <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKd65f.6eecb976.0;received=92.63.138.100 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-721f To: sip:92.63.138.97:5060;tag=as4a7a774b Call-ID: 6ad64986-7670@92.63.138.100 CSeq: 10 INFO Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK71cc1a2d;rport=5060 Record-Route: From: "C9486-drainageex-london" ;tag=as7f4ad4f6 To: ;tag=S9672ZrHt475F Call-ID: 01faa6fa6a0303774a75319231829daa@sipipgw.magrathea.net CSeq: 102 INVITE Contact: User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Min-SE: 120 Content-Type: application/sdp Content-Length: 181 v=0 o=- 38578177 0 IN IP4 213.166.5.153 s=Cisco SDP 0 c=IN IP4 213.166.5.153 t=0 0 m=audio 41138 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (14 headers 9 lines) --- node4*CLI> list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.148, port 5060 Transmitting (no NAT) to 213.166.5.148:5060: ACK sip:213.166.5.153 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK38997448;rport Route: Max-Forwards: 70 From: "C9486-drainageex-london" ;tag=as7f4ad4f6 To: ;tag=S9672ZrHt475F Contact: Call-ID: 01faa6fa6a0303774a75319231829daa@sipipgw.magrathea.net CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "C9486-drainageex-london" ;privacy=off;screen=yes Content-Length: 0 --- -- SIP/magrathea-outbound-0a1bd1f8 answered SIP/213.166.5.140-b6635528 node4*CLI> Reliably Transmitting (no NAT) to 213.166.5.148:5060: OPTIONS sip:sipipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK50cdc621;rport Max-Forwards: 70 From: "asterisk" ;tag=as59d6f75a To: Contact: Call-ID: 25e4bbe60c2c32cd4d6d4da542094a1f@92.63.138.97 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.1.3-rc1 Date: Tue, 29 Sep 2009 15:54:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 403 OPTIONS not supported Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK50cdc621;rport=5060 From: "asterisk" ;tag=as59d6f75a To: ;tag=9a264c9a00f926193bf7ce80aab147c3.b321 Call-ID: 25e4bbe60c2c32cd4d6d4da542094a1f@92.63.138.97 CSeq: 102 OPTIONS Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- node4*CLI> Really destroying SIP dialog '25e4bbe60c2c32cd4d6d4da542094a1f@92.63.138.97' Method: OPTIONS node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INFO sip:92.63.138.97:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKaa5f.367baf11.0 To: sip:92.63.138.97:5060 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-cc60 CSeq: 10 INFO Call-ID: 6ad6498a-7670@92.63.138.100 Content-Length: 0 User-Agent: OpenSIPS (1.5.1-notls (i386/linux)) <-------------> --- (8 headers 0 lines) --- node4*CLI> <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKaa5f.367baf11.0;received=92.63.138.100 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-cc60 To: sip:92.63.138.97:5060;tag=as6f5b1b31 Call-ID: 6ad6498a-7670@92.63.138.100 CSeq: 10 INFO Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> BYE sip:07540693656@92.63.138.97 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 213.166.5.148;branch=z9hG4bK1c73.f0de8eb4.0 Via: SIP/2.0/UDP 213.166.5.139:5070;branch=z9hG4bK1c73.f2a9d982.0 Via: SIP/2.0/UDP 213.166.5.153;rport=5060;branch=z9hG4bKtcmyUHDaZc4Xa Max-Forwards: 68 From: ;tag=UUrgDXrX3jQ2g To: "Number-1" ;tag=as06213673 Call-ID: 499ec2950521784b11c6325a1c1bacfb@sipipgw.magrathea.net CSeq: 121001503 BYE User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Sending to 213.166.5.148 : 5060 (no NAT) node4*CLI> <--- Transmitting (no NAT) to 213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 213.166.5.148;branch=z9hG4bK1c73.f0de8eb4.0;received=213.166.5.148 Via: SIP/2.0/UDP 213.166.5.139:5070;branch=z9hG4bK1c73.f2a9d982.0 Via: SIP/2.0/UDP 213.166.5.153;rport=5060;branch=z9hG4bKtcmyUHDaZc4Xa Record-Route: From: ;tag=UUrgDXrX3jQ2g To: "Number-1" ;tag=as06213673 Call-ID: 499ec2950521784b11c6325a1c1bacfb@sipipgw.magrathea.net CSeq: 121001503 BYE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> -- Executing [h@insight-dialout-external:1] Set("SIP/213.166.5.140-b6608498", "CDR(outbound)=02070934488") in new stack node4*CLI> == Spawn extension (insight-dialout-external, 442033931863, 5) exited non-zero on 'SIP/213.166.5.140-b6608498' node4*CLI> Scheduling destruction of SIP dialog 'F33839A5-AC4611DE-AE77D31F-EFD16753@213.166.5.140' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 92.63.138.100, port 5060 Reliably Transmitting (no NAT) to 92.63.138.100:5060: BYE sip:07540693656@213.166.5.140:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK1be0537c;rport Route: , Max-Forwards: 70 From: ;tag=as7079980e To: "07540693656" ;tag=C672D7AC-1238 Call-ID: F33839A5-AC4611DE-AE77D31F-EFD16753@213.166.5.140 CSeq: 102 BYE User-Agent: Asterisk PBX 1.6.1.3-rc1 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- node4*CLI> == MixMonitor close filestream node4*CLI> == End MixMonitor Recording SIP/213.166.5.140-b6608498 node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK1be0537c;rport=5060 From: ;tag=as7079980e To: ;tag=C672D7AC-1238 Date: Tue, 29 Sep 2009 15:54:40 gmt Call-ID: F33839A5-AC4611DE-AE77D31F-EFD16753@213.166.5.140 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 BYE Reason: Q.850;cause=16 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- node4*CLI> SIP Response message for INCOMING dialog BYE arrived node4*CLI> Really destroying SIP dialog 'F33839A5-AC4611DE-AE77D31F-EFD16753@213.166.5.140' Method: ACK node4*CLI> Really destroying SIP dialog '499ec2950521784b11c6325a1c1bacfb@sipipgw.magrathea.net' Method: BYE node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INFO sip:92.63.138.97:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK6a5f.1268bfd2.0 To: sip:92.63.138.97:5060 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-7a36 CSeq: 10 INFO Call-ID: 6ad6498e-7670@92.63.138.100 Content-Length: 0 User-Agent: OpenSIPS (1.5.1-notls (i386/linux)) <-------------> --- (8 headers 0 lines) --- node4*CLI> <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK6a5f.1268bfd2.0;received=92.63.138.100 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-7a36 To: sip:92.63.138.97:5060;tag=as0e611ee3 Call-ID: 6ad6498e-7670@92.63.138.100 CSeq: 10 INFO Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INVITE sip:448447041612@92.63.138.97:5060 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKee6a.f58ea6d3.0 Via: SIP/2.0/UDP 87.238.72.153;branch=z9hG4bKee6a.29394306.0 Via: SIP/2.0/UDP 87.238.72.155:5060;branch=z9hG4bK1FD2D398E From: "01314416818" ;tag=A5D81F40-B1D To: node4*CLI> Date: Tue, 29 Sep 2009 15:55:08 gmt Call-ID: 4C38ED4F-AC4711DE-9D26BA8E-F0DAC5A2@87.238.72.155 Supported: timer Min-SE: 1800 User-Agent: MSSGW(B) Allow: INVITE, BYE, CANCEL, ACK CSeq: 101 INVITE Max-Forwards: 13 Remote-Party-ID: ;party=calling;screen=yes;privacy=off Timestamp: 1254239708 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 404 v=0 o=CiscoSystemsSIP-GW-UserAgent 5593 5499 IN IP4 87.238.72.155 s=SIP Call c=IN IP4 87.238.72.155 t=0 0 m=audio 17698 RTP/AVP 8 18 4 3 2 0 101 c=IN IP4 87.238.72.155 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:3 GSM/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (23 headers 17 lines) --- node4*CLI> == Using SIP RTP CoS mark 5 Sending to 92.63.138.100 : 5060 (no NAT) Using INVITE request as basis request - 4C38ED4F-AC4711DE-9D26BA8E-F0DAC5A2@87.238.72.155 No matching peer for '01314416818' from '92.63.138.100:5060' Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 3 Found RTP audio format 2 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 87.238.72.155:17698 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Got unsupported a:fmtp in SDP offer Found audio description format G723 for ID 4 Got unsupported a:fmtp in SDP offer Found audio description format GSM for ID 3 Found audio description format G726-32 for ID 2 Found audio description format PCMU for ID 0 node4*CLI> Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8 (alaw), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 87.238.72.155:17698 Looking for 448447041612 in incoming_magrathea (domain 92.63.138.97) node4*CLI> list_route: hop: list_route: hop: node4*CLI> <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKee6a.f58ea6d3.0;received=92.63.138.100 Via: SIP/2.0/UDP 87.238.72.153;branch=z9hG4bKee6a.29394306.0 Via: SIP/2.0/UDP 87.238.72.155:5060;branch=z9hG4bK1FD2D398E Record-Route: Record-Route: From: "01314416818" ;tag=A5D81F40-B1D To: Call-ID: 4C38ED4F-AC4711DE-9D26BA8E-F0DAC5A2@87.238.72.155 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> node4*CLI> -- Executing AGI("SIP/87.238.72.155-b660d258", "agi://web0/track.agi?username=emsinternet&campaignName=Website+General&campaignId=204&destination=01925413333&salesTracking=yes&recordCall=yes&analytics=yes") node4*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERPRES()=allowed) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(accountcode)=emsinternet) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(userfield)=204) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERID(name)=Website General) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(amaflags)=billing) node4*CLI> -- AGI Script agi://web0/track.agi?username=emsinternet&campaignName=Website+General&campaignId=204&destination=01925413333&salesTracking=yes&recordCall=yes&analytics=yes completed, returning 0 node4*CLI> -- Executing [448447041612@insight-dialout-external:1] GotoIf("SIP/87.238.72.155-b660d258", "1?2:5") in new stack -- Goto (insight-dialout-external,448447041612,2) -- Executing [448447041612@insight-dialout-external:2] Answer("SIP/87.238.72.155-b660d258", "") in new stack node4*CLI> Audio is at 92.63.138.97 port 12352 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKee6a.f58ea6d3.0;received=92.63.138.100 Via: SIP/2.0/UDP 87.238.72.153;branch=z9hG4bKee6a.29394306.0 Via: SIP/2.0/UDP 87.238.72.155:5060;branch=z9hG4bK1FD2D398E Record-Route: Record-Route: From: "01314416818" ;tag=A5D81F40-B1D To: ;tag=as0e692e17 Call-ID: 4C38ED4F-AC4711DE-9D26BA8E-F0DAC5A2@87.238.72.155 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 267 v=0 o=root 1249950457 1249950457 IN IP4 92.63.138.97 s=Asterisk PBX 1.6.1.3-rc1 c=IN IP4 92.63.138.97 t=0 0 m=audio 12352 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> ACK sip:448447041612@92.63.138.97:5060 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKee6a.f58ea6d3.2 Via: SIP/2.0/UDP 87.238.72.153;branch=z9hG4bKee6a.29394306.2 Via: SIP/2.0/UDP 87.238.72.155:5060;branch=z9hG4bK1FD2D41880 From: ;tag=A5D81F40-B1D To: ;tag=as0e692e17 D node4*CLI> ate: Tue, 29 Sep 2009 15:55:08 gmt Call-ID: 4C38ED4F-AC4711DE-9D26BA8E-F0DAC5A2@87.238.72.155 Max-Forwards: 13 CSeq: 101 ACK Content-Length: 0 <-------------> --- (13 headers 0 lines) --- node4*CLI> -- Executing [448447041612@insight-dialout-external:3] MixMonitor("SIP/87.238.72.155-b660d258", "voip2-1254239708.659.wav") in new stack node4*CLI> -- Executing [448447041612@insight-dialout-external:4] BackGround("SIP/87.238.72.155-b660d258", "adinsight-call-recorded") in new stack node4*CLI> == Begin MixMonitor Recording SIP/87.238.72.155-b660d258 node4*CLI> -- Playing 'adinsight-call-recorded.ulaw' (language 'en') node4*CLI> -- Executing [448447041612@insight-dialout-external:5] Dial("SIP/87.238.72.155-b660d258", "SIP/magrathea-outbound/01925413333,40,CrF(trackSales^448447041612:204:voip2-1254239708.659:01925413333^1)") in new stack node4*CLI> == Using SIP RTP CoS mark 5 node4*CLI> Audio is at 92.63.138.97 port 14328 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP node4*CLI> Reliably Transmitting (no NAT) to 213.166.5.148:5060: INVITE sip:01925413333@sipipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK2d95b8c6;rport Max-Forwards: 70 From: "Website General" ;tag=as5f701e7e To: Contact: Call-ID: 4aead2f93854c1955bc59cd054e584b7@sipipgw.magrathea.net CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "Website General" ;privacy=off;screen=yes Date: Tue, 29 Sep 2009 15:55:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 265 v=0 o=root 901420057 901420057 IN IP4 92.63.138.97 s=Asterisk PBX 1.6.1.3-rc1 c=IN IP4 92.63.138.97 t=0 0 m=audio 14328 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- node4*CLI> -- Called magrathea-outbound/01925413333 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK2d95b8c6;rport=5060 From: "Website General" ;tag=as5f701e7e To: Call-ID: 4aead2f93854c1955bc59cd054e584b7@sipipgw.magrathea.net CSeq: 102 INVITE Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK2d95b8c6;rport=5060 Record-Route: From: "Website General" ;tag=as5f701e7e To: ;tag=pBF28aF438p6K Call-ID: 4aead2f93854c1955bc59cd054e584b7@sipipgw.magrathea.net CSeq: 102 INVITE Contact: User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Type: application/sdp Content-Length: 181 v=0 o=- 38579132 0 IN IP4 213.166.5.154 s=Cisco SDP 0 c=IN IP4 213.166.5.154 t=0 0 m=audio 31442 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (13 headers 9 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 213.166.5.154:31442 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 213.166.5.154:31442 node4*CLI> -- SIP/magrathea-outbound-0a160f28 is making progress passing it to SIP/87.238.72.155-b660d258 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK2d95b8c6;rport=5060 Record-Route: From: "Website General" ;tag=as5f701e7e To: ;tag=pBF28aF438p6K Call-ID: 4aead2f93854c1955bc59cd054e584b7@sipipgw.magrathea.net CSeq: 102 INVITE Contact: User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Min-SE: 120 Content-Type: application/sdp Content-Length: 181 v=0 o=- 38579132 0 IN IP4 213.166.5.154 s=Cisco SDP 0 c=IN IP4 213.166.5.154 t=0 0 m=audio 31442 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (14 headers 9 lines) --- list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.148, port 5060 Transmitting (no NAT) to 213.166.5.148:5060: ACK sip:213.166.5.154 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK707a3043;rport Route: Max-Forwards: 70 From: "Website General" ;tag=as5f701e7e To: ;tag=pBF28aF438p6K Contact: Call-ID: 4aead2f93854c1955bc59cd054e584b7@sipipgw.magrathea.net CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "Website General" ;privacy=off;screen=yes Content-Length: 0 --- node4*CLI> -- SIP/magrathea-outbound-0a160f28 answered SIP/87.238.72.155-b660d258 node4*CLI> Reliably Transmitting (no NAT) to 213.166.5.148:5060: OPTIONS sip:sipipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK4b6d7721;rport Max-Forwards: 70 From: "asterisk" ;tag=as4f56ab31 To: Contact: Call-ID: 1b09386c07229c58082b1cb30567ba77@92.63.138.97 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.1.3-rc1 Date: Tue, 29 Sep 2009 15:55:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 403 OPTIONS not supported Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK4b6d7721;rport=5060 From: "asterisk" ;tag=as4f56ab31 To: ;tag=9a264c9a00f926193bf7ce80aab147c3.03f7 Call-ID: 1b09386c07229c58082b1cb30567ba77@92.63.138.97 CSeq: 102 OPTIONS Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '1b09386c07229c58082b1cb30567ba77@92.63.138.97' Method: OPTIONS node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INFO sip:92.63.138.97:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK054f.befe6041.0 To: sip:92.63.138.97:5060 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-023a CSeq: 10 INFO Call-ID: 6ad64992-7670@92.63.138.100 Content-Length: 0 User-Agent: OpenSIPS (1.5.1-notls (i386/linux)) <-------------> --- (8 headers 0 lines) --- <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK054f.befe6041.0;received=92.63.138.100 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-023a To: sip:92.63.138.97:5060;tag=as20490dee Call-ID: 6ad64992-7670@92.63.138.100 CSeq: 10 INFO Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INVITE sip:448005244604@92.63.138.97:5060 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKe53.248f4ff6.0 Via: SIP/2.0/UDP 87.238.72.153;branch=z9hG4bKe53.11283a37.0 Via: SIP/2.0/UDP 87.238.72.155:5060;branch=z9hG4bK1FD2FC13DD From: "07944992213" ;tag=A5D87E58-C19 To: D node4*CLI> ate: Tue, 29 Sep 2009 15:55:33 gmt Call-ID: 5ABAED89-AC4711DE-9D85BA8E-F0DAC5A2@87.238.72.155 Supported: timer Min-SE: 1800 User-Agent: MSSGW(B) Allow: INVITE, BYE, CANCEL, ACK CSeq: 101 INVITE Max-Forwards: 13 Remote-Party-ID: ;party=calling;screen=yes;privacy=off Timestamp: 1254239733 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 404 v=0 o=CiscoSystemsSIP-GW-UserAgent 2102 2697 IN IP4 87.238.72.155 s=SIP Call c=IN IP4 87.238.72.155 t=0 0 m=audio 19290 RTP/AVP 8 18 4 3 2 0 101 c=IN IP4 87.238.72.155 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:3 GSM/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (23 headers 17 lines) --- == Using SIP RTP CoS mark 5 Sending to 92.63.138.100 : 5060 (no NAT) Using INVITE request as basis request - 5ABAED89-AC4711DE-9D85BA8E-F0DAC5A2@87.238.72.155 No matching peer for '07944992213' from '92.63.138.100:5060' Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 3 Found RTP audio format 2 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 87.238.72.155:19290 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Got unsupported a:fmtp in SDP offer Found audio description format G723 for ID 4 Got unsupported a:fmtp in SDP offer Found audio description format GSM for ID 3 Found audio description format G726-32 for ID 2 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8 (alaw), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 87.238.72.155:19290 Looking for 448005244604 in incoming_magrathea (domain 92.63.138.97) node4*CLI> list_route: hop: list_route: hop: <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKe53.248f4ff6.0;received=92.63.138.100 Via: SIP/2.0/UDP 87.238.72.153;branch=z9hG4bKe53.11283a37.0 Via: SIP/2.0/UDP 87.238.72.155:5060;branch=z9hG4bK1FD2FC13DD Record-Route: Record-Route: From: "07944992213" ;tag=A5D87E58-C19 To: Call-ID: 5ABAED89-AC4711DE-9D85BA8E-F0DAC5A2@87.238.72.155 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> node4*CLI> -- Executing AGI("SIP/87.238.72.155-b6648770", "agi://web0/track.agi?username=provisioning&campaignName=Number-1&campaignId=1151&destination=02031373322&recordCall=yes") node4*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERPRES()=allowed) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(accountcode)=provisioning) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(userfield)=1151) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERID(name)=Number-1) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(amaflags)=billing) node4*CLI> -- AGI Script agi://web0/track.agi?username=provisioning&campaignName=Number-1&campaignId=1151&destination=02031373322&recordCall=yes completed, returning 0 node4*CLI> -- Executing [448005244604@insight-dialout-external:1] GotoIf("SIP/87.238.72.155-b6648770", "1?2:5") in new stack node4*CLI> -- Goto (insight-dialout-external,448005244604,2) -- Executing [448005244604@insight-dialout-external:2] Answer("SIP/87.238.72.155-b6648770", "") in new stack node4*CLI> Audio is at 92.63.138.97 port 19368 node4*CLI> Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP node4*CLI> <--- Reliably Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKe53.248f4ff6.0;received=92.63.138.100 Via: SIP/2.0/UDP 87.238.72.153;branch=z9hG4bKe53.11283a37.0 Via: SIP/2.0/UDP 87.238.72.155:5060;branch=z9hG4bK1FD2FC13DD Record-Route: Record-Route: From: "07944992213" ;tag=A5D87E58-C19 To: ;tag=as4b57d28d Call-ID: 5ABAED89-AC4711DE-9D85BA8E-F0DAC5A2@87.238.72.155 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 267 v=0 o=root 1384575926 1384575926 IN IP4 92.63.138.97 s=Asterisk PBX 1.6.1.3-rc1 c=IN IP4 92.63.138.97 t=0 0 m=audio 19368 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> ACK sip:448005244604@92.63.138.97:5060 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKe53.248f4ff6.2 Via: SIP/2.0/UDP 87.238.72.153;branch=z9hG4bKe53.11283a37.2 Via: SIP/2.0/UDP 87.238.72.155:5060;branch=z9hG4bK1FD2FD1D3 From: ;tag=A5D87E58-C19 To: ;tag=as4b57d28d Date: Tue, 29 Sep 2009 15:55:33 gmt Call-ID: 5ABAED89-AC4711DE-9D85BA8E-F0DAC5A2@87.238.72.155 Max-Forwards: 13 CSeq: 101 ACK Content-Length: 0 <-------------> --- (13 headers 0 lines) --- node4*CLI> -- Executing [448005244604@insight-dialout-external:3] MixMonitor("SIP/87.238.72.155-b6648770", "voip2-1254239733.661.wav") in new stack node4*CLI> -- Executing [448005244604@insight-dialout-external:4] BackGround("SIP/87.238.72.155-b6648770", "adinsight-call-recorded") in new stack == Begin MixMonitor Recording SIP/87.238.72.155-b6648770 node4*CLI> -- Playing 'adinsight-call-recorded.ulaw' (language 'en') node4*CLI> -- Executing [448005244604@insight-dialout-external:5] Dial("SIP/87.238.72.155-b6648770", "SIP/magrathea-outbound/02031373322,40,Cr") in new stack node4*CLI> == Using SIP RTP CoS mark 5 node4*CLI> Audio is at 92.63.138.97 port 19766 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP node4*CLI> Reliably Transmitting (no NAT) to 213.166.5.148:5060: INVITE sip:02031373322@sipipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK3702330e;rport Max-Forwards: 70 From: "Number-1" ;tag=as1a5d5655 To: Contact: Call-ID: 6df7b37f4776e04b6e4d450b09f7f3d6@sipipgw.magrathea.net CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "Number-1" ;privacy=off;screen=yes Date: Tue, 29 Sep 2009 15:55:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 267 v=0 o=root 1300720890 1300720890 IN IP4 92.63.138.97 s=Asterisk PBX 1.6.1.3-rc1 c=IN IP4 92.63.138.97 t=0 0 m=audio 19766 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- node4*CLI> -- Called magrathea-outbound/02031373322 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK3702330e;rport=5060 From: "Number-1" ;tag=as1a5d5655 To: Call-ID: 6df7b37f4776e04b6e4d450b09f7f3d6@sipipgw.magrathea.net CSeq: 102 INVITE Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK3702330e;rport=5060 Record-Route: From: "Number-1" ;tag=as1a5d5655 To: ;tag=tB319ZFN08SeN Call-ID: 6df7b37f4776e04b6e4d450b09f7f3d6@sipipgw.magrathea.net CSeq: 102 INVITE Contact: User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Type: application/sdp Content-Length: 229 v=0 o=- 1254236291 1254236291 IN IP4 213.166.5.143 s=- c=IN IP4 213.166.5.143 t=0 0 m=audio 24054 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sqn: 0 a=cdsc: 1 image udptl t38 <-------------> --- (13 headers 11 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 213.166.5.143:24054 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 213.166.5.143:24054 node4*CLI> -- SIP/magrathea-outbound-0a18ca58 is making progress passing it to SIP/87.238.72.155-b6648770 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK3702330e;rport=5060 Record-Route: From: "Number-1" ;tag=as1a5d5655 To: ;tag=tB319ZFN08SeN Call-ID: 6df7b37f4776e04b6e4d450b09f7f3d6@sipipgw.magrathea.net CSeq: 102 INVITE Contact: User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Min-SE: 120 Content-Type: application/sdp Content-Length: 229 v=0 o=- 1254236291 1254236291 IN IP4 213.166.5.143 s=- c=IN IP4 213.166.5.143 t=0 0 m=audio 24054 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sqn: 0 a=cdsc: 1 image udptl t38 <-------------> --- (14 headers 11 lines) --- list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.148, port 5060 Transmitting (no NAT) to 213.166.5.148:5060: ACK sip:213.166.5.153 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK23162cb2;rport Route: Max-Forwards: 70 From: "Number-1" ;tag=as1a5d5655 To: ;tag=tB319ZFN08SeN Contact: Call-ID: 6df7b37f4776e04b6e4d450b09f7f3d6@sipipgw.magrathea.net CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "Number-1" ;privacy=off;screen=yes Content-Length: 0 --- node4*CLI> -- SIP/magrathea-outbound-0a18ca58 answered SIP/87.238.72.155-b6648770 node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INFO sip:92.63.138.97:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKc44f.5d6fcfb5.0 To: sip:92.63.138.97:5060 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-b46c CSeq: 10 INFO Call-ID: 6ad64996-7670@92.63.138.100 Content-Length: 0 User-Agent: OpenSIPS (1.5.1-notls (i386/linux)) <-------------> --- (8 headers 0 lines) --- <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKc44f.5d6fcfb5.0;received=92.63.138.100 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-b46c To: sip:92.63.138.97:5060;tag=as60563fe8 Call-ID: 6ad64996-7670@92.63.138.100 CSeq: 10 INFO Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> Reliably Transmitting (no NAT) to 213.166.5.148:5060: OPTIONS sip:sipipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK599cce3f;rport Max-Forwards: 70 From: "asterisk" ;tag=as2d8a8cd1 To: Contact: Call-ID: 395754c32ec84ac404960374469e87ea@92.63.138.97 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.1.3-rc1 Date: Tue, 29 Sep 2009 15:56:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 403 OPTIONS not supported Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK599cce3f;rport=5060 From: "asterisk" ;tag=as2d8a8cd1 To: ;tag=9a264c9a00f926193bf7ce80aab147c3.4d95 Call-ID: 395754c32ec84ac404960374469e87ea@92.63.138.97 CSeq: 102 OPTIONS Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '395754c32ec84ac404960374469e87ea@92.63.138.97' Method: OPTIONS node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INFO sip:92.63.138.97:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK984f.091c3a17.0 To: sip:92.63.138.97:5060 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-0a13 CSeq: 10 INFO Call-ID: 6ad6499a-7670@92.63.138.100 Content-Length: 0 User-Agent: OpenSIPS (1.5.1-notls (i386/linux)) <-------------> --- (8 headers 0 lines) --- <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK984f.091c3a17.0;received=92.63.138.100 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-0a13 To: sip:92.63.138.97:5060;tag=as15a59ed7 Call-ID: 6ad6499a-7670@92.63.138.100 CSeq: 10 INFO Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> BYE sip:448081201310@92.63.138.97:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKbe4d.000b5f44.0 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bKbe4d.5d4448e.0 Via: SIP/2.0/UDP 213.166.5.140:5060;branch=z9hG4bK5048871A0A From: ;tag=C673CEF8-12CE To: ;tag=as16c295b7 Date: Tue, 29 Sep 2009 15:53:42 gmt Call-ID: 18F448DF-AC4711DE-B0B1D31F-EFD16753@213.166.5.140 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 13 Timestamp: 1254239794 CSeq: 102 BYE Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Sending to 92.63.138.100 : 5060 (no NAT) <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKbe4d.000b5f44.0;received=92.63.138.100 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bKbe4d.5d4448e.0 Via: SIP/2.0/UDP 213.166.5.140:5060;branch=z9hG4bK5048871A0A Record-Route: From: ;tag=C673CEF8-12CE To: ;tag=as16c295b7 Call-ID: 18F448DF-AC4711DE-B0B1D31F-EFD16753@213.166.5.140 CSeq: 102 BYE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> -- Executing [h@insight-dialout-external:1] Set("SIP/213.166.5.140-b6635528", "CDR(outbound)=01425475777") in new stack node4*CLI> Scheduling destruction of SIP dialog '01faa6fa6a0303774a75319231829daa@sipipgw.magrathea.net' in 6400 ms (Method: INVITE) node4*CLI> set_destination: Parsing for address/port to send to node4*CLI> set_destination: set destination to 213.166.5.148, port 5060 node4*CLI> Reliably Transmitting (no NAT) to 213.166.5.148:5060: BYE sip:213.166.5.153 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK486976de;rport Route: Max-Forwards: 70 From: "C9486-drainageex-london" ;tag=as7f4ad4f6 To: ;tag=S9672ZrHt475F Call-ID: 01faa6fa6a0303774a75319231829daa@sipipgw.magrathea.net CSeq: 103 BYE User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "C9486-drainageex-london" ;privacy=off;screen=yes X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- node4*CLI> == Spawn extension (insight-dialout-external, 448081201310, 5) exited non-zero on 'SIP/213.166.5.140-b6635528' node4*CLI> == MixMonitor close filestream == End MixMonitor Recording SIP/213.166.5.140-b6635528 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK486976de;rport=5060 From: "C9486-drainageex-london" ;tag=as7f4ad4f6 To: ;tag=S9672ZrHt475F Call-ID: 01faa6fa6a0303774a75319231829daa@sipipgw.magrathea.net CSeq: 103 BYE User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Length: 0 <-------------> --- (10 headers 0 lines) --- node4*CLI> Really destroying SIP dialog '01faa6fa6a0303774a75319231829daa@sipipgw.magrathea.net' Method: INVITE Really destroying SIP dialog '18F448DF-AC4711DE-B0B1D31F-EFD16753@213.166.5.140' Method: BYE node4*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 92.63.138.100, port 5060 node4*CLI> Audio is at 92.63.138.97 port 11888 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP node4*CLI> Reliably Transmitting (no NAT) to 92.63.138.100:5060: INVITE sip:07971296525@213.166.5.140:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK5d387261;rport Route: , Max-Forwards: 70 From: ;tag=as598e3492 To: "07971296525" ;tag=C668B120-1FDD Contact: Call-ID: 66B6355D-AC4511DE-9169D31F-EFD16753@213.166.5.140 C node4*CLI> Seq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.3-rc1 Require: timer Session-Expires: 1800;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer X-asterisk-Info: SIP re-invite (Session-Timers) Content-Type: application/sdp Content-Length: 267 v=0 o=root 1025224544 1025224544 IN IP4 92.63.138.97 s=Asterisk PBX 1.6.1.3-rc1 c=IN IP4 92.63.138.97 t=0 0 m=audio 11888 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK5d387261;rport=5060 From: ;tag=as598e3492 To: "07971296525" ;tag=C668B120-1FDD Call-ID: 66B6355D-AC4511DE-9169D31F-EFD16753@213.166.5.140 CSeq: 102 INVITE Server: AdInsight SIP Balancer Content-Length: 0 <-------------> --- (8 headers 0 lines) --- node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK5d387261;rport=5060 From: ;tag=as598e3492 To: "07971296525" ;tag=C668B120-1FDD Call-ID: 66B6355D-AC4511DE-9169D31F-EFD16753@213.166.5.140 CSeq: 102 INVITE Server: Asterisk PBX 1.6.1.0-rc4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 <-------------> --- (10 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 92.63.138.100, port 5060 Transmitting (no NAT) to 92.63.138.100:5060: ACK sip:07971296525@213.166.5.140:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK5d387261;rport Route: , Max-Forwards: 70 From: ;tag=as598e3492 To: "07971296525" ;tag=C668B120-1FDD Contact: Call-ID: 66B6355D-AC4511DE-9169D31F-EFD16753@213.166.5.140 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.3-rc1 Content-Length: 0 --- -- Executing [h@insight-dialout-external:1] Set("SIP/213.166.5.140-b7b80688", "CDR(outbound)=07841131297") in new stack node4*CLI> == Spawn extension (insight-dialout-external, 441616607985, 5) exited non-zero on 'SIP/213.166.5.140-b7b80688' -- Executing [441616607985:2681:voip2-1254238894.629:07841131297@trackSales:1] AGI("SIP/magrathea-outbound-b6619730", "agi://web0/adinsight.agi") in new stack node4*CLI> == MixMonitor close filestream == End MixMonitor Recording SIP/213.166.5.140-b7b80688 node4*CLI> -- Playing 'beep' (escape_digits=) (sample_offset 0) node4*CLI> -- Playing 'adinsight-menu.ulaw' (language 'en') node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INVITE sip:448005244621@92.63.138.97:5060 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK92a6.85de63f3.0 Via: SIP/2.0/UDP 213.166.5.129;branch=z9hG4bK92a6.9b2291e7.0 Via: SIP/2.0/UDP 213.166.5.133:5060;branch=z9hG4bK57A09111CE7 From: "02085315529" ;tag=65DF1480-1731 To: Date: Tue, 29 Sep 2009 15:56:36 gmt Call-ID: 80839FEF-AC4711DE-9570EFB9-710EBE4B@213.166.5.133 Supported: timer,replaces Min-SE: 1800 User-Agent: MSSGW Allow: INVITE, BYE, CANCEL, ACK CSeq: 101 INVITE Max-Forwards: 13 Remote-Party-ID: ;party=calling;screen=yes;privacy=off Timestamp: 1254239796 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 403 v=0 o=CiscoSystemsSIP-GW-UserAgent 1674 569 IN IP4 213.166.5.133 s=SIP Call c=IN IP4 213.166.5.133 t=0 0 m=audio 19248 RTP/AVP 8 18 4 3 2 0 101 c=IN IP4 213.166.5.133 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:3 GSM/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> node4*CLI> --- (23 headers 17 lines) --- node4*CLI> == Using SIP RTP CoS mark 5 node4*CLI> Sending to 92.63.138.100 : 5060 (no NAT) Using INVITE request as basis request - 80839FEF-AC4711DE-9570EFB9-710EBE4B@213.166.5.133 No matching peer for '02085315529' from '92.63.138.100:5060' Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 3 Found RTP audio format 2 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 213.166.5.133:19248 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Got unsupported a:fmtp in SDP offer node4*CLI> Found audio description format G723 for ID 4 Got unsupported a:fmtp in SDP offer Found audio description format GSM for ID 3 Found audio description format G726-32 for ID 2 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8 (alaw), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 213.166.5.133:19248 Looking for 448005244621 in incoming_magrathea (domain 92.63.138.97) node4*CLI> list_route: hop: list_route: hop: node4*CLI> <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK92a6.85de63f3.0;received=92.63.138.100 Via: SIP/2.0/UDP 213.166.5.129;branch=z9hG4bK92a6.9b2291e7.0 Via: SIP/2.0/UDP 213.166.5.133:5060;branch=z9hG4bK57A09111CE7 Record-Route: Record-Route: From: "02085315529" ;tag=65DF1480-1731 To: Call-ID: 80839FEF-AC4711DE-9570EFB9-710EBE4B@213.166.5.133 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> node4*CLI> -- Executing AGI("SIP/213.166.5.133-b7b894a8", "agi://web0/track.agi?username=freestart office&campaignName=sales+freephone&campaignId=838&destination=01942406100&recordCall=yes") node4*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERPRES()=allowed) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(accountcode)=freestart office) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(userfield)=838) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERID(name)=sales freephone) node4*CLI> -- AGI Script Executing Application: (Set) Options: (CDR(amaflags)=billing) node4*CLI> -- AGI Script agi://web0/track.agi?username=freestart office&campaignName=sales+freephone&campaignId=838&destination=01942406100&recordCall=yes completed, returning 0 node4*CLI> -- Executing [448005244621@insight-dialout-external:1] GotoIf("SIP/213.166.5.133-b7b894a8", "1?2:5") in new stack node4*CLI> -- Goto (insight-dialout-external,448005244621,2) node4*CLI> -- Executing [448005244621@insight-dialout-external:2] Answer("SIP/213.166.5.133-b7b894a8", "") in new stack node4*CLI> Audio is at 92.63.138.97 port 19804 node4*CLI> Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP node4*CLI> <--- Reliably Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK92a6.85de63f3.0;received=92.63.138.100 Via: SIP/2.0/UDP 213.166.5.129;branch=z9hG4bK92a6.9b2291e7.0 Via: SIP/2.0/UDP 213.166.5.133:5060;branch=z9hG4bK57A09111CE7 Record-Route: Record-Route: From: "02085315529" ;tag=65DF1480-1731 To: ;tag=as7be0add2 Call-ID: 80839FEF-AC4711DE-9570EFB9-710EBE4B@213.166.5.133 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 267 v=0 o=root 1349310246 1349310246 IN IP4 92.63.138.97 s=Asterisk PBX 1.6.1.3-rc1 c=IN IP4 92.63.138.97 t=0 0 m=audio 19804 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> ACK sip:448005244621@92.63.138.97:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK92a6.85de63f3.2 Via: SIP/2.0/UDP 213.166.5.129;branch=z9hG4bK92a6.9b2291e7.2 Via: SIP/2.0/UDP 213.166.5.133:5060;branch=z9hG4bK57A0912100D From: ;tag=65DF1480-1731 To: ;tag=as7be0add2 Date: Tue, 29 Sep 2009 15:56:36 gmt Call-ID: 80839FEF-AC4711DE-9570EFB9-710EBE4B@213.166.5.133 Max-Forwards: 13 CSeq: 101 ACK Content-Length: 0 <-------------> --- (12 headers 0 lines) --- node4*CLI> -- Executing [448005244621@insight-dialout-external:3] MixMonitor("SIP/213.166.5.133-b7b894a8", "voip2-1254239796.663.wav") in new stack node4*CLI> -- Executing [448005244621@insight-dialout-external:4] BackGround("SIP/213.166.5.133-b7b894a8", "adinsight-call-recorded") in new stack == Begin MixMonitor Recording SIP/213.166.5.133-b7b894a8 node4*CLI> -- Playing 'adinsight-call-recorded.ulaw' (language 'en') node4*CLI> -- Executing [448005244621@insight-dialout-external:5] Dial("SIP/213.166.5.133-b7b894a8", "SIP/magrathea-outbound/01942406100,40,Cr") in new stack node4*CLI> == Using SIP RTP CoS mark 5 node4*CLI> Audio is at 92.63.138.97 port 15668 node4*CLI> Adding codec 0x8 (alaw) to SDP node4*CLI> Adding non-codec 0x1 (telephone-event) to SDP node4*CLI> Reliably Transmitting (no NAT) to 213.166.5.148:5060: INVITE sip:01942406100@sipipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK42d991b1;rport Max-Forwards: 70 From: "sales freephone" ;tag=as152c0410 To: Contact: Call-ID: 0db5cdb23daa7b710c9b3b333a479d2e@sipipgw.magrathea.net CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "sales freephone" ;privacy=off;screen=yes Date: Tue, 29 Sep 2009 15:56:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 265 v=0 o=root 400755491 400755491 IN IP4 92.63.138.97 s=Asterisk PBX 1.6.1.3-rc1 c=IN IP4 92.63.138.97 t=0 0 m=audio 15668 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- node4*CLI> -- Called magrathea-outbound/01942406100 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK42d991b1;rport=5060 From: "sales freephone" ;tag=as152c0410 To: Call-ID: 0db5cdb23daa7b710c9b3b333a479d2e@sipipgw.magrathea.net CSeq: 102 INVITE Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- node4*CLI> [Sep 29 16:56:41] NOTICE[14783]: rtp.c:1135 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 87.238.72.155 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK42d991b1;rport=5060 Record-Route: From: "sales freephone" ;tag=as152c0410 To: ;tag=4p7eXNvBt3ZpF Call-ID: 0db5cdb23daa7b710c9b3b333a479d2e@sipipgw.magrathea.net CSeq: 102 INVITE Contact: User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Type: application/sdp Content-Length: 181 v=0 o=- 42094388 0 IN IP4 213.166.5.154 s=Cisco SDP 0 c=IN IP4 213.166.5.154 t=0 0 m=audio 33514 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (13 headers 9 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 213.166.5.154:33514 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 213.166.5.154:33514 node4*CLI> -- SIP/magrathea-outbound-0a1bd1f8 is making progress passing it to SIP/213.166.5.133-b7b894a8 node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK42d991b1;rport=5060 Record-Route: From: "sales freephone" ;tag=as152c0410 To: ;tag=4p7eXNvBt3ZpF Call-ID: 0db5cdb23daa7b710c9b3b333a479d2e@sipipgw.magrathea.net CSeq: 102 INVITE Contact: User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Min-SE: 120 Content-Type: application/sdp Content-Length: 181 v=0 o=- 42094388 0 IN IP4 213.166.5.154 s=Cisco SDP 0 c=IN IP4 213.166.5.154 t=0 0 m=audio 33514 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (14 headers 9 lines) --- list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.148, port 5060 Transmitting (no NAT) to 213.166.5.148:5060: ACK sip:213.166.5.154 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK28ce21cc;rport Route: Max-Forwards: 70 From: "sales freephone" ;tag=as152c0410 To: ;tag=4p7eXNvBt3ZpF Contact: Call-ID: 0db5cdb23daa7b710c9b3b333a479d2e@sipipgw.magrathea.net CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "sales freephone" ;privacy=off;screen=yes Content-Length: 0 --- node4*CLI> -- SIP/magrathea-outbound-0a1bd1f8 answered SIP/213.166.5.133-b7b894a8 node4*CLI> -- AGI Script agi://web0/adinsight.agi completed, returning 0 -- Executing [441616607985:2681:voip2-1254238894.629:07841131297@trackSales:2] AGI("SIP/magrathea-outbound-b6619730", "agi://web0/adinsight.agi") in new stack node4*CLI> -- Playing 'beep' (escape_digits=) (sample_offset 0) node4*CLI> -- Playing 'adinsight-menu.ulaw' (language 'en') node4*CLI> <--- SIP read from UDP://92.63.138.100:5060 ---> INFO sip:92.63.138.97:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK584f.b40d83f4.0 To: sip:92.63.138.97:5060 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-bc45 CSeq: 10 INFO Call-ID: 6ad6499e-7670@92.63.138.100 Content-Length: 0 User-Agent: OpenSIPS (1.5.1-notls (i386/linux)) <-------------> --- (8 headers 0 lines) --- <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK584f.b40d83f4.0;received=92.63.138.100 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-bc45 To: sip:92.63.138.97:5060;tag=as6bed47b8 Call-ID: 6ad6499e-7670@92.63.138.100 CSeq: 10 INFO Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> -- AGI Script agi://web0/adinsight.agi completed, returning 0 -- Auto fallthrough, channel 'SIP/magrathea-outbound-b6619730' status is 'UNKNOWN' -- Executing [h@trackSales:1] Set("SIP/magrathea-outbound-b6619730", "CDR(outbound)=07841131297") in new stack Scheduling destruction of SIP dialog '388b46b059088c105173be163249b34e@sipipgw.magrathea.net' in 6400 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.148, port 5060 Reliably Transmitting (no NAT) to 213.166.5.148:5060: BYE sip:87.238.72.134 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK0bb888c5;rport Route: Max-Forwards: 70 From: "fault test" ;tag=as44f78ecd To: ;tag=H74SUjgXeQ88r Call-ID: 388b46b059088c105173be163249b34e@sipipgw.magrathea.net CSeq: 103 BYE User-Agent: Asterisk PBX 1.6.1.3-rc1 Remote-Party-ID: "fault test" ;privacy=off;screen=yes X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- node4*CLI> <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.138.97:5060;received=92.63.138.97;branch=z9hG4bK0bb888c5;rport=5060 From: "fault test" ;tag=as44f78ecd To: ;tag=H74SUjgXeQ88r Call-ID: 388b46b059088c105173be163249b34e@sipipgw.magrathea.net CSeq: 103 BYE User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '388b46b059088c105173be163249b34e@sipipgw.magrathea.net' Method: INVITE node4*CLI> quit Reliably Transmitting (no NAT) to 213.166.5.148:5060: OPTIONS sip:sipipgw.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK488596d0;rport Max-Forwards: 70 From: "asterisk" ;tag=as2ee3e30d To: Contact: Call-ID: 5f9fcbfd5b253a381793727160c32b4d@92.63.138.97 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.1.3-rc1 Date: Tue, 29 Sep 2009 15:57:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- node4*CLI> quit <--- SIP read from UDP://213.166.5.148:5060 ---> SIP/2.0 403 OPTIONS not supported Via: SIP/2.0/UDP 92.63.138.97:5060;branch=z9hG4bK488596d0;rport=5060 From: "asterisk" ;tag=as2ee3e30d To: ;tag=9a264c9a00f926193bf7ce80aab147c3.c11c Call-ID: 5f9fcbfd5b253a381793727160c32b4d@92.63.138.97 CSeq: 102 OPTIONS Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '5f9fcbfd5b253a381793727160c32b4d@92.63.138.97' Method: OPTIONS node4*CLI> quit <--- SIP read from UDP://92.63.138.100:5060 ---> INFO sip:92.63.138.97:5060 SIP/2.0 Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK8532.dd9fb9b5.0 To: sip:92.63.138.97:5060 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-f7dc CSeq: 10 INFO Call-ID: 6ad649a2-7670@92.63.138.100 Content-Length: 0 User-Agent: OpenSIPS (1.5.1-notls (i386/linux)) <-------------> --- (8 headers 0 lines) --- <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bK8532.dd9fb9b5.0;received=92.63.138.100 From: ;tag=d0a287e22b4acdc4f33a4c2d24646723-f7dc To: sip:92.63.138.97:5060;tag=as24ba52e6 Call-ID: 6ad649a2-7670@92.63.138.100 CSeq: 10 INFO Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> quit <--- SIP read from UDP://92.63.138.100:5060 ---> BYE sip:441616607985@92.63.138.97:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKad5c.63bcf965.0 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bKad5c.66eba5e2.0 Via: SIP/2.0/UDP 213.166.5.140:5060;branch=z9hG4bK504947520 From: ;tag=C668B120-1FDD To: ;tag=as598e3492 Date: Tue, 29 Sep 2009 15:41:34 gmt Call-ID: 66B6355D-AC4511DE-9169D31F-EFD16753@213.166.5.140 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 13 Timestamp: 1254239861 CSeq: 102 BYE Content-Length: 0 <-------------> --- (14 headers 0 lines) --- <--- Transmitting (no NAT) to 92.63.138.100:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 92.63.138.100;branch=z9hG4bKad5c.63bcf965.0;received=92.63.138.100 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bKad5c.66eba5e2.0 Via: SIP/2.0/UDP 213.166.5.140:5060;branch=z9hG4bK504947520 From: ;tag=C668B120-1FDD To: ;tag=as598e3492 Call-ID: 66B6355D-AC4511DE-9169D31F-EFD16753@213.166.5.140 CSeq: 102 BYE Server: Asterisk PBX 1.6.1.3-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> node4*CLI> uquit