Script started on Wed Sep 3 12:00:38 2008 [1;32m[[0;36mpbx1.myhostcenter.com[1;34m:[0;36m~[1;32m] [0;34m[1;34m([0mroot[0m[1;34m)[1;33m>[0mexitasterisk -vvvvvvvvvr Asterisk 1.4.20-1 RPM by vc-rpms@voipconsulting.nl, Copyright (C) 1999 - 2008 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. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': Found Connected to Asterisk 1.4.20-1 RPM by vc-rpms@voipconsulting.nl currently running on pbx1 (pid = 23164) pbx1*CLI> Verbosity is at least 27 [Kpbx1*CLI> [Sep 3 12:02:00] <--- SIP read from 69.211.143.171:5063 ---> <-------------> [Kpbx1*CLI> [Sep 3 12:02:00] <--- SIP read from 69.211.143.171:5064 ---> <-------------> [Kpbx1*CLI> [Sep 3 12:02:03] <--- SIP read from 213.91.247.166:20228 ---> INVITE sip:2712@pbx1.myhostcenter.com SIP/2.0 Via: SIP/2.0/UDP 213.91.247.166:20228;branch=z9hG4bK-d8754z-05794957623b4f70-1---d8754z-;rport Max-Forwards: 70 Contact: To: "Andy Mentges" From: ;tag=f650aa2c Call-ID: Y2JjMGI3MTU4M2YzZWY3ZDQ0M2U4NDc2MDAyYTBlZTI. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1100l stamp 47546 Content-Length: 483 v=0 o=- 8 2 IN IP4 192.168.10.7 s=CounterPath X-Lite 3.0 c=IN IP4 213.91.247.166 t=0 0 m=audio 34426 RTP/AVP 107 119 100 106 6 0 97 105 98 8 102 3 5 101 a=alt:1 1 : tb+RK+ZS 8YC4G+wO 192.168.10.7 34426 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:119 BV32-FEC/16000 a=rtpmap:100 SPEEX/16000 a=rtpmap:106 SPEEX-FEC/16000 a=rtpmap:97 SPEEX/8000 a=rtpmap:105 SPEEX-FEC/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:102 L16/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> [Sep 3 12:02:03] --- (12 headers 18 lines) --- [Sep 3 12:02:03] Sending to 213.91.247.166 : 20228 (NAT) [Sep 3 12:02:03] Using INVITE request as basis request - Y2JjMGI3MTU4M2YzZWY3ZDQ0M2U4NDc2MDAyYTBlZTI. [Sep 3 12:02:03] <--- Reliably Transmitting (NAT) to 213.91.247.166:20228 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 213.91.247.166:20228;branch=z9hG4bK-d8754z-05794957623b4f70-1---d8754z-;received=213.91.247.166;rport=20228 From: ;tag=f650aa2c To: "Andy Mentges";tag=as5321397e Call-ID: Y2JjMGI3MTU4M2YzZWY3ZDQ0M2U4NDc2MDAyYTBlZTI. CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1d15d0da" Content-Length: 0 <------------> [Sep 3 12:02:03] Scheduling destruction of SIP dialog 'Y2JjMGI3MTU4M2YzZWY3ZDQ0M2U4NDc2MDAyYTBlZTI.' in 32000 ms (Method: INVITE) [Sep 3 12:02:03] Found user '2720' [Kpbx1*CLI> [Sep 3 12:02:04] <--- SIP read from 213.91.247.166:20228 ---> ACK sip:2712@pbx1.myhostcenter.com SIP/2.0 Via: SIP/2.0/UDP 213.91.247.166:20228;branch=z9hG4bK-d8754z-05794957623b4f70-1---d8754z-;rport To: "Andy Mentges";tag=as5321397e From: ;tag=f650aa2c Call-ID: Y2JjMGI3MTU4M2YzZWY3ZDQ0M2U4NDc2MDAyYTBlZTI. CSeq: 1 ACK Content-Length: 0 <-------------> [Sep 3 12:02:04] --- (7 headers 0 lines) --- [Kpbx1*CLI> [Sep 3 12:02:04] <--- SIP read from 213.91.247.166:20228 ---> INVITE sip:2712@pbx1.myhostcenter.com SIP/2.0 Via: SIP/2.0/UDP 213.91.247.166:20228;branch=z9hG4bK-d8754z-8710190ab47adb2d-1---d8754z-;rport Max-Forwards: 70 Contact: To: "Andy Mentges" From: ;tag=f650aa2c Call-ID: Y2JjMGI3MTU4M2YzZWY3ZDQ0M2U4NDc2MDAyYTBlZTI. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Proxy-Authorization: Digest username="2720",realm="asterisk",nonce="1d15d0da",uri="sip:2712@pbx1.myhostcenter.com",response="72c4c4a78cd87001e37bdcb249a1032a",algorithm=MD5 User-Agent: X-Lite release 1100l stamp 47546 Content-Length: 483 v=0 o=- 8 2 IN IP4 192.168.10.7 s=CounterPath X-Lite 3.0 c=IN IP4 213.91.247.166 t=0 0 m=audio 34426 RTP/AVP 107 119 100 106 6 0 97 105 98 8 102 3 5 101 a=alt:1 1 : tb+RK+ZS 8YC4G+wO 192.168.10.7 34426 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:119 BV32-FEC/16000 a=rtpmap:100 SPEEX/16000 a=rtpmap:106 SPEEX-FEC/16000 a=rtpmap:97 SPEEX/8000 a=rtpmap:105 SPEEX-FEC/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:102 L16/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> [Sep 3 12:02:04] --- (13 headers 18 lines) --- [Sep 3 12:02:04] Sending to 213.91.247.166 : 20228 (NAT) [Sep 3 12:02:04] Using INVITE request as basis request - Y2JjMGI3MTU4M2YzZWY3ZDQ0M2U4NDc2MDAyYTBlZTI. [Sep 3 12:02:04] Found user '2720' [Sep 3 12:02:04] Found RTP audio format 107 [Sep 3 12:02:04] Found RTP audio format 119 [Sep 3 12:02:04] Found RTP audio format 100 [Sep 3 12:02:04] Found RTP audio format 106 [Sep 3 12:02:04] Found RTP audio format 6 [Sep 3 12:02:04] Found RTP audio format 0 [Sep 3 12:02:04] Found RTP audio format 97 [Sep 3 12:02:04] Found RTP audio format 105 [Sep 3 12:02:04] Found RTP audio format 98 [Sep 3 12:02:04] Found RTP audio format 8 [Sep 3 12:02:04] Found RTP audio format 102 [Sep 3 12:02:04] Found RTP audio format 3 [Sep 3 12:02:04] Found RTP audio format 5 [Sep 3 12:02:04] Found RTP audio format 101 [Sep 3 12:02:04] Peer audio RTP is at port 213.91.247.166:34426 [Sep 3 12:02:04] Found unknown media description format BV32 for ID 107 [Sep 3 12:02:04] Found unknown media description format BV32-FEC for ID 119 [Sep 3 12:02:04] Found audio description format SPEEX for ID 100 [Sep 3 12:02:04] Found unknown media description format SPEEX-FEC for ID 106 [Sep 3 12:02:04] Found audio description format SPEEX for ID 97 [Sep 3 12:02:04] Found unknown media description format SPEEX-FEC for ID 105 [Sep 3 12:02:04] Found audio description format iLBC for ID 98 [Sep 3 12:02:04] Found audio description format L16 for ID 102 [Sep 3 12:02:04] Found audio description format telephone-event for ID 101 [Sep 3 12:02:04] Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x66e (gsm|ulaw|alaw|adpcm|slin|speex|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) [Sep 3 12:02:04] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Sep 3 12:02:04] Peer audio RTP is at port 213.91.247.166:34426 [Sep 3 12:02:04] Looking for 2712 in from-internal (domain pbx1.myhostcenter.com) [Sep 3 12:02:04] list_route: hop: [Sep 3 12:02:04] <--- Transmitting (NAT) to 213.91.247.166:20228 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 213.91.247.166:20228;branch=z9hG4bK-d8754z-8710190ab47adb2d-1---d8754z-;received=213.91.247.166;rport=20228 From: ;tag=f650aa2c To: "Andy Mentges" Call-ID: Y2JjMGI3MTU4M2YzZWY3ZDQ0M2U4NDc2MDAyYTBlZTI. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Kpbx1*CLI> [Sep 3 12:02:04] -- Executing [2712@from-internal:1] GotoIf("SIP/2720-0933bac0", "1?ext-local|2712|1") in new stack [Sep 3 12:02:04] -- Goto (ext-local,2712,1) [Sep 3 12:02:04] -- Executing [2712@ext-local:1] Macro("SIP/2720-0933bac0", "exten-vm|2712|2712") in new stack [Sep 3 12:02:04] -- Executing [s@macro-exten-vm:1] Macro("SIP/2720-0933bac0", "user-callerid") in new stack [Sep 3 12:02:04] -- Executing [s@macro-user-callerid:1] NoOp("SIP/2720-0933bac0", "user-callerid: device 2720") in new stack [Sep 3 12:02:04] -- Executing [s@macro-user-callerid:2] Set("SIP/2720-0933bac0", "AMPUSER=2720") in new stack [Sep 3 12:02:04] -- Executing [s@macro-user-callerid:3] GotoIf("SIP/2720-0933bac0", "0?report") in new stack [Sep 3 12:02:04] -- Executing [s@macro-user-callerid:4] ExecIf("SIP/2720-0933bac0", "1|Set|REALCALLERIDNUM=2720") in new stack [Sep 3 12:02:04] -- Executing [s@macro-user-callerid:5] NoOp("SIP/2720-0933bac0", "REALCALLERIDNUM is 2720") in new stack [Sep 3 12:02:04] -- Executing [s@macro-user-callerid:6] Set("SIP/2720-0933bac0", "AMPUSER=2720") in new stack [Sep 3 12:02:04] -- Executing [s@macro-user-callerid:7] Set("SIP/2720-0933bac0", "AMPUSERCIDNAME=Stefan Dimitrov") in new stack [Sep 3 12:02:04] -- Executing [s@macro-user-callerid:8] GotoIf("SIP/2720-0933bac0", "0?report") in new stack [Sep 3 12:02:04] -- Executing [s@macro-user-callerid:9] Set("SIP/2720-0933bac0", "AMPUSERCID=2720") in new stack [Sep 3 12:02:04] -- Executing [s@macro-user-callerid:10] Set("SIP/2720-0933bac0", "CALLERID(all)="Stefan Dimitrov" <2720>") in new stack [Sep 3 12:02:04] -- Executing [s@macro-user-callerid:11] Set("SIP/2720-0933bac0", "REALCALLERIDNUM=2720") in new stack [Sep 3 12:02:04] -- Executing [s@macro-user-callerid:12] ExecIf("SIP/2720-0933bac0", "0|Set|CHANNEL(language)=") in new stack [Sep 3 12:02:04] -- Executing [s@macro-user-callerid:13] NoOp("SIP/2720-0933bac0", "TTL: ARG1: 2712") in new stack [Sep 3 12:02:04] -- Executing [s@macro-user-callerid:14] GotoIf("SIP/2720-0933bac0", "0?continue") in new stack [Sep 3 12:02:04] -- Executing [s@macro-user-callerid:15] Set("SIP/2720-0933bac0", "__TTL=64") in new stack [Sep 3 12:02:04] -- Executing [s@macro-user-callerid:16] GotoIf("SIP/2720-0933bac0", "1?continue") in new stack [Sep 3 12:02:04] -- Goto (macro-user-callerid,s,23) [Sep 3 12:02:04] -- Executing [s@macro-user-callerid:23] NoOp("SIP/2720-0933bac0", "Using CallerID "Stefan Dimitrov" <2720>") in new stack [Sep 3 12:02:04] -- Executing [s@macro-exten-vm:2] Set("SIP/2720-0933bac0", "RingGroupMethod=none") in new stack [Sep 3 12:02:04] -- Executing [s@macro-exten-vm:3] Set("SIP/2720-0933bac0", "VMBOX=2712") in new stack [Sep 3 12:02:04] -- Executing [s@macro-exten-vm:4] Set("SIP/2720-0933bac0", "EXTTOCALL=2712") in new stack [Sep 3 12:02:04] -- Executing [s@macro-exten-vm:5] Set("SIP/2720-0933bac0", "CFUEXT=") in new stack [Sep 3 12:02:04] -- Executing [s@macro-exten-vm:6] Set("SIP/2720-0933bac0", "CFBEXT=") in new stack [Sep 3 12:02:04] -- Executing [s@macro-exten-vm:7] Set("SIP/2720-0933bac0", "RT=15") in new stack [Sep 3 12:02:04] -- Executing [s@macro-exten-vm:8] Macro("SIP/2720-0933bac0", "record-enable|2712|IN") in new stack [Sep 3 12:02:04] -- Executing [s@macro-record-enable:1] GotoIf("SIP/2720-0933bac0", "0?2:4") in new stack [Sep 3 12:02:04] -- Goto (macro-record-enable,s,4) [Sep 3 12:02:04] -- Executing [s@macro-record-enable:4] AGI("SIP/2720-0933bac0", "recordingcheck|20080903-120204|1220457724.6") in new stack [Sep 3 12:02:04] -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck [Kpbx1*CLI> [Sep 3 12:02:04] recordingcheck|20080903-120204|1220457724.6: Inbound recording enabled. [Kpbx1*CLI> [Sep 3 12:02:04] recordingcheck|20080903-120204|1220457724.6: CALLFILENAME=20080903-120204-1220457724.6 [Kpbx1*CLI> [Sep 3 12:02:04] -- AGI Script recordingcheck completed, returning 0 [Kpbx1*CLI> [Sep 3 12:02:04] -- Executing [s@macro-record-enable:999] MixMonitor("SIP/2720-0933bac0", "20080903-120204-1220457724.6.wav") in new stack [Sep 3 12:02:04] -- Executing [s@macro-exten-vm:9] Macro("SIP/2720-0933bac0", "dial|15|tr|2712") in new stack [Sep 3 12:02:04] -- Executing [s@macro-dial:1] GotoIf("SIP/2720-0933bac0", "1?dial") in new stack [Sep 3 12:02:04] -- Goto (macro-dial,s,3) [Sep 3 12:02:04] -- Executing [s@macro-dial:3] AGI("SIP/2720-0933bac0", "dialparties.agi") in new stack [Kpbx1*CLI> [Sep 3 12:02:04] == Begin MixMonitor Recording SIP/2720-0933bac0 [Sep 3 12:02:04] -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi [Kpbx1*CLI> [Sep 3 12:02:04] dialparties.agi: Starting New Dialparties.agi [Kpbx1*CLI> [Sep 3 12:02:04] == Parsing '/etc/asterisk/manager.conf': [Sep 3 12:02:04] Found [Kpbx1*CLI> [Sep 3 12:02:04] == Parsing '/etc/asterisk/manager_additional.conf': [Sep 3 12:02:04] Found [Kpbx1*CLI> [Sep 3 12:02:04] == Parsing '/etc/asterisk/manager_custom.conf': [Sep 3 12:02:04] Found [Kpbx1*CLI> [Sep 3 12:02:04] == Manager 'admin' logged on from 127.0.0.1 [Kpbx1*CLI> [Sep 3 12:02:04] dialparties.agi: Caller ID name is 'Stefan Dimitrov' number is '2720' [Kpbx1*CLI> [Sep 3 12:02:04] dialparties.agi: USE_CONFIRMATION: 'FALSE' [Kpbx1*CLI> [Sep 3 12:02:04] dialparties.agi: RINGGROUP_INDEX: '' [Kpbx1*CLI> [Sep 3 12:02:04] dialparties.agi: Methodology of ring is 'none' [Kpbx1*CLI> [Sep 3 12:02:04] -- dialparties.agi: Added extension 2712 to extension map [Kpbx1*CLI> [Sep 3 12:02:04] -- dialparties.agi: Extension 2712 cf is disabled [Kpbx1*CLI> [Sep 3 12:02:04] -- dialparties.agi: Extension 2712 do not disturb is disabled [Kpbx1*CLI> [Sep 3 12:02:04] > dialparties.agi: extnum 2712 has: cw: 0; hascfb: 0 [] hascfu: 0 [] [Kpbx1*CLI> [Sep 3 12:02:04] > dialparties.agi: ExtensionState: 0 [Kpbx1*CLI> [Sep 3 12:02:04] dialparties.agi: Extension 2712 has ExtensionState: 0 [Kpbx1*CLI> [Sep 3 12:02:04] -- dialparties.agi: Checking CW and CFB status for extension 2712 [Kpbx1*CLI> [Sep 3 12:02:04] -- dialparties.agi: dbset CALLTRACE/2712 to 2720 [Kpbx1*CLI> [Sep 3 12:02:04] -- dialparties.agi: Filtered ARG3: 2712 [Kpbx1*CLI> [Sep 3 12:02:04] == Manager 'admin' logged off from 127.0.0.1 [Kpbx1*CLI> [Sep 3 12:02:04] -- AGI Script dialparties.agi completed, returning 0 [Kpbx1*CLI> [Sep 3 12:02:04] -- Executing [s@macro-dial:7] Dial("SIP/2720-0933bac0", "SIP/2712|15|tr") in new stack [Kpbx1*CLI> [Sep 3 12:02:04] Audio is at 64.6.240.50 port 12508 [Sep 3 12:02:04] Adding codec 0x4 (ulaw) to SDP [Sep 3 12:02:04] Adding codec 0x8 (alaw) to SDP [Sep 3 12:02:04] Adding non-codec 0x1 (telephone-event) to SDP [Sep 3 12:02:04] Reliably Transmitting (no NAT) to 65.112.77.100:5060: INVITE sip:2712@65.112.77.100 SIP/2.0 Via: SIP/2.0/UDP 64.6.240.50:5060;branch=z9hG4bK6ba6c56a;rport From: "Stefan Dimitrov" ;tag=as229109c5 To: Contact: Call-ID: 1ac6c6825371e82004e4df45558f8aba@64.6.240.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 03 Sep 2008 16:02:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 262 v=0 o=root 23164 23164 IN IP4 64.6.240.50 s=session c=IN IP4 64.6.240.50 t=0 0 m=audio 12508 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Sep 3 12:02:04] -- Called 2712 [Sep 3 12:02:04] <--- Transmitting (NAT) to 213.91.247.166:20228 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 213.91.247.166:20228;branch=z9hG4bK-d8754z-8710190ab47adb2d-1---d8754z-;received=213.91.247.166;rport=20228 From: ;tag=f650aa2c To: "Andy Mentges";tag=as28cef870 Call-ID: Y2JjMGI3MTU4M2YzZWY3ZDQ0M2U4NDc2MDAyYTBlZTI. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Kpbx1*CLI> [Sep 3 12:02:04] <--- SIP read from 65.112.77.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 64.6.240.50:5060;branch=z9hG4bK6ba6c56a;rport From: "Stefan Dimitrov" ;tag=as229109c5 To: ;tag=8F1E8693-401864E2 CSeq: 102 INVITE Call-ID: 1ac6c6825371e82004e4df45558f8aba@64.6.240.50 Contact: User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.3.0401 Content-Length: 0 <-------------> [Sep 3 12:02:04] --- (9 headers 0 lines) --- [Kpbx1*CLI> [Sep 3 12:02:04] <--- SIP read from 65.112.77.100:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 64.6.240.50:5060;branch=z9hG4bK6ba6c56a;rport From: "Stefan Dimitrov" ;tag=as229109c5 To: ;tag=8F1E8693-401864E2 CSeq: 102 INVITE Call-ID: 1ac6c6825371e82004e4df45558f8aba@64.6.240.50 Contact: User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.3.0401 Allow-Events: talk,hold,conference Content-Length: 0 <-------------> [Sep 3 12:02:04] --- (10 headers 0 lines) --- [Kpbx1*CLI> [Sep 3 12:02:04] -- SIP/2712-0932cef0 is ringing [Kpbx1*CLI> [Sep 3 12:02:06] <--- SIP read from 65.112.77.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 64.6.240.50:5060;branch=z9hG4bK6ba6c56a;rport From: "Stefan Dimitrov" ;tag=as229109c5 To: ;tag=8F1E8693-401864E2 CSeq: 102 INVITE Call-ID: 1ac6c6825371e82004e4df45558f8aba@64.6.240.50 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.3.0401 Content-Type: application/sdp Content-Length: 201 v=0 o=- 1220457375 1220457375 IN IP4 65.112.77.100 s=Polycom IP Phone c=IN IP4 65.112.77.100 t=0 0 m=audio 2244 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 <-------------> [Sep 3 12:02:06] --- (11 headers 9 lines) --- [Sep 3 12:02:06] Found RTP audio format 0 [Sep 3 12:02:06] Found RTP audio format 101 [Sep 3 12:02:06] Peer audio RTP is at port 65.112.77.100:2244 [Sep 3 12:02:06] Found audio description format PCMU for ID 0 [Sep 3 12:02:06] Found audio description format telephone-event for ID 101 [Sep 3 12:02:06] Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Sep 3 12:02:06] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Sep 3 12:02:06] Peer audio RTP is at port 65.112.77.100:2244 [Sep 3 12:02:06] list_route: hop: [Sep 3 12:02:06] set_destination: Parsing for address/port to send to [Sep 3 12:02:06] set_destination: set destination to 65.112.77.100, port 5060 [Sep 3 12:02:06] Transmitting (no NAT) to 65.112.77.100:5060: ACK sip:2712@65.112.77.100 SIP/2.0 Via: SIP/2.0/UDP 64.6.240.50:5060;branch=z9hG4bK1ee4e11d;rport From: "Stefan Dimitrov" ;tag=as229109c5 To: ;tag=8F1E8693-401864E2 Contact: Call-ID: 1ac6c6825371e82004e4df45558f8aba@64.6.240.50 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Kpbx1*CLI> [Sep 3 12:02:06] -- SIP/2712-0932cef0 answered SIP/2720-0933bac0 [Sep 3 12:02:06] Audio is at 64.6.240.50 port 12662 [Sep 3 12:02:06] Adding codec 0x4 (ulaw) to SDP [Sep 3 12:02:06] Adding codec 0x8 (alaw) to SDP [Sep 3 12:02:06] Adding non-codec 0x1 (telephone-event) to SDP [Sep 3 12:02:06] <--- Reliably Transmitting (NAT) to 213.91.247.166:20228 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 213.91.247.166:20228;branch=z9hG4bK-d8754z-8710190ab47adb2d-1---d8754z-;received=213.91.247.166;rport=20228 From: ;tag=f650aa2c To: "Andy Mentges";tag=as28cef870 Call-ID: Y2JjMGI3MTU4M2YzZWY3ZDQ0M2U4NDc2MDAyYTBlZTI. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 262 v=0 o=root 23164 23164 IN IP4 64.6.240.50 s=session c=IN IP4 64.6.240.50 t=0 0 m=audio 12662 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Kpbx1*CLI> [Sep 3 12:02:07] Retransmitting #1 (NAT) to 213.91.247.166:20228: SIP/2.0 200 OK Via: SIP/2.0/UDP 213.91.247.166:20228;branch=z9hG4bK-d8754z-8710190ab47adb2d-1---d8754z-;received=213.91.247.166;rport=20228 From: ;tag=f650aa2c To: "Andy Mentges";tag=as28cef870 Call-ID: Y2JjMGI3MTU4M2YzZWY3ZDQ0M2U4NDc2MDAyYTBlZTI. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 262 v=0 o=root 23164 23164 IN IP4 64.6.240.50 s=session c=IN IP4 64.6.240.50 t=0 0 m=audio 12662 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Kpbx1*CLI> [Sep 3 12:02:08] Retransmitting #2 (NAT) to 213.91.247.166:20228: SIP/2.0 200 OK Via: SIP/2.0/UDP 213.91.247.166:20228;branch=z9hG4bK-d8754z-8710190ab47adb2d-1---d8754z-;received=213.91.247.166;rport=20228 From: ;tag=f650aa2c To: "Andy Mentges";tag=as28cef870 Call-ID: Y2JjMGI3MTU4M2YzZWY3ZDQ0M2U4NDc2MDAyYTBlZTI. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 262 v=0 o=root 23164 23164 IN IP4 64.6.240.50 s=session c=IN IP4 64.6.240.50 t=0 0 m=audio 12662 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Kpbx1*CLI> [Sep 3 12:02:10] Retransmitting #3 (NAT) to 213.91.247.166:20228: SIP/2.0 200 OK Via: SIP/2.0/UDP 213.91.247.166:20228;branch=z9hG4bK-d8754z-8710190ab47adb2d-1---d8754z-;received=213.91.247.166;rport=20228 From: ;tag=f650aa2c To: "Andy Mentges";tag=as28cef870 Call-ID: Y2JjMGI3MTU4M2YzZWY3ZDQ0M2U4NDc2MDAyYTBlZTI. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 262 v=0 o=root 23164 23164 IN IP4 64.6.240.50 s=session c=IN IP4 64.6.240.50 t=0 0 m=audio 12662 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Kpbx1*CLI> [Sep 3 12:02:10] <--- SIP read from 69.211.143.171:5063 ---> <-------------> [Kpbx1*CLI> [Sep 3 12:02:10] <--- SIP read from 69.211.143.171:5064 ---> <-------------> [Kpbx1*CLI> [Sep 3 12:02:14] Retransmitting #4 (NAT) to 213.91.247.166:20228: SIP/2.0 200 OK Via: SIP/2.0/UDP 213.91.247.166:20228;branch=z9hG4bK-d8754z-8710190ab47adb2d-1---d8754z-;received=213.91.247.166;rport=20228 From: ;tag=f650aa2c To: "Andy Mentges";tag=as28cef870 Call-ID: Y2JjMGI3MTU4M2YzZWY3ZDQ0M2U4NDc2MDAyYTBlZTI. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 262 v=0 o=root 23164 23164 IN IP4 64.6.240.50 s=session c=IN IP4 64.6.240.50 t=0 0 m=audio 12662 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Kpbx1*CLI> [Sep 3 12:02:17] Really destroying SIP dialog 'YTg5NzkxZmM3OGNkYmI1OWZhZjZkOGEyNWFlNWI5YWE.' Method: REGISTER [Kpbx1*CLI> [Sep 3 12:02:18] Retransmitting #5 (NAT) to 213.91.247.166:20228: SIP/2.0 200 OK Via: SIP/2.0/UDP 213.91.247.166:20228;branch=z9hG4bK-d8754z-8710190ab47adb2d-1---d8754z-;received=213.91.247.166;rport=20228 From: ;tag=f650aa2c To: "Andy Mentges";tag=as28cef870 Call-ID: Y2JjMGI3MTU4M2YzZWY3ZDQ0M2U4NDc2MDAyYTBlZTI. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 262 v=0 o=root 23164 23164 IN IP4 64.6.240.50 s=session c=IN IP4 64.6.240.50 t=0 0 m=audio 12662 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Kpbx1*CLI> [Sep 3 12:02:20] <--- SIP read from 69.211.143.171:5063 ---> <-------------> [Kpbx1*CLI> [Sep 3 12:02:20] <--- SIP read from 69.211.143.171:5064 ---> <-------------> [Kpbx1*CLI> [Sep 3 12:02:20] <--- SIP read from 213.91.247.166:20228 ---> <-------------> [Kpbx1*CLI> [Sep 3 12:02:21] Reliably Transmitting (NAT) to 64.6.243.44:5060: OPTIONS sip:2731@64.6.243.44 SIP/2.0 Via: SIP/2.0/UDP 64.6.240.50:5060;branch=z9hG4bK625c91ad;rport From: "Unknown" ;tag=as25f7bbff To: Contact: Call-ID: 4407d6ac449746741800de764813b861@64.6.240.50 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 03 Sep 2008 16:02:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Kpbx1*CLI> [Sep 3 12:02:21] <--- SIP read from 64.6.243.44:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 64.6.240.50:5060;branch=z9hG4bK625c91ad;rport From: "Unknown" ;tag=as25f7bbff To: ;tag=5CEB3FB2-8B02E449 CSeq: 102 OPTIONS Call-ID: 4407d6ac449746741800de764813b861@64.6.240.50 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 Content-Length: 0 <-------------> [Sep 3 12:02:21] --- (10 headers 0 lines) --- [Kpbx1*CLI> [Sep 3 12:02:21] Really destroying SIP dialog '4407d6ac449746741800de764813b861@64.6.240.50' Method: OPTIONS [Kpbx1*CLI> [Sep 3 12:02:22] Retransmitting #6 (NAT) to 213.91.247.166:20228: SIP/2.0 200 OK Via: SIP/2.0/UDP 213.91.247.166:20228;branch=z9hG4bK-d8754z-8710190ab47adb2d-1---d8754z-;received=213.91.247.166;rport=20228 From: ;tag=f650aa2c To: "Andy Mentges";tag=as28cef870 Call-ID: Y2JjMGI3MTU4M2YzZWY3ZDQ0M2U4NDc2MDAyYTBlZTI. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 262 v=0 o=root 23164 23164 IN IP4 64.6.240.50 s=session c=IN IP4 64.6.240.50 t=0 0 m=audio 12662 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Kpbx1*CLI> [Sep 3 12:02:22] Reliably Transmitting (NAT) to 75.118.140.183:5060: OPTIONS sip:2798@192.168.0.50 SIP/2.0 Via: SIP/2.0/UDP 64.6.240.50:5060;branch=z9hG4bK507ed336;rport From: "Unknown" ;tag=as2485d389 To: Contact: Call-ID: 3ba09719030a657655574be242745e35@64.6.240.50 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 03 Sep 2008 16:02:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Kpbx1*CLI> [Sep 3 12:02:22] Reliably Transmitting (NAT) to 24.131.113.117:55002: OPTIONS sip:2799@192.168.2.10 SIP/2.0 Via: SIP/2.0/UDP 64.6.240.50:5060;branch=z9hG4bK658a95bc;rport From: "Unknown" ;tag=as5797c74b To: Contact: Call-ID: 04e43c651448b96b3da3c55e6cc266fe@64.6.240.50 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 03 Sep 2008 16:02:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Kpbx1*CLI> [Sep 3 12:02:22] <--- SIP read from 75.118.140.183:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 64.6.240.50:5060;branch=z9hG4bK507ed336;rport From: "Unknown" ;tag=as2485d389 To: ;tag=9D0368E9-6FF366EC CSeq: 102 OPTIONS Call-ID: 3ba09719030a657655574be242745e35@64.6.240.50 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_301-UA/3.0.3.0401 Content-Length: 0 <-------------> [Sep 3 12:02:22] --- (10 headers 0 lines) --- [Kpbx1*CLI> [Sep 3 12:02:22] Really destroying SIP dialog '3ba09719030a657655574be242745e35@64.6.240.50' Method: OPTIONS [Kpbx1*CLI> [Sep 3 12:02:22] <--- SIP read from 24.131.113.117:55002 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 64.6.240.50:5060;branch=z9hG4bK658a95bc;rport From: "Unknown" ;tag=as5797c74b To: ;tag=6FAACDD0-13A77657 CSeq: 102 OPTIONS Call-ID: 04e43c651448b96b3da3c55e6cc266fe@64.6.240.50 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.3.0401 Content-Length: 0 <-------------> [Sep 3 12:02:22] --- (10 headers 0 lines) --- [Kpbx1*CLI> [Sep 3 12:02:22] Really destroying SIP dialog '04e43c651448b96b3da3c55e6cc266fe@64.6.240.50' Method: OPTIONS [Kpbx1*CLI> [Sep 3 12:02:23] Really destroying SIP dialog 'NDI5MDhmZjVmMzMxMTIyN2NmMjA2MGJhZTZjZmFmMTk.' Method: REGISTER [Kpbx1*CLI> [Sep 3 12:02:23] Reliably Transmitting (NAT) to 96.231.70.93:50910: OPTIONS sip:2723@192.168.1.159 SIP/2.0 Via: SIP/2.0/UDP 64.6.240.50:5060;branch=z9hG4bK4bd1152b;rport From: "Unknown" ;tag=as6042f228 To: Contact: Call-ID: 1dc0b2b65c2a793a5ea477821e21b511@64.6.240.50 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 03 Sep 2008 16:02:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Kpbx1*CLI> [Sep 3 12:02:23] <--- SIP read from 96.231.70.93:50910 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 64.6.240.50:5060;branch=z9hG4bK4bd1152b;rport From: "Unknown" ;tag=as6042f228 To: ;tag=57E68532-64E9F3C1 CSeq: 102 OPTIONS Call-ID: 1dc0b2b65c2a793a5ea477821e21b511@64.6.240.50 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.3.0401 Content-Length: 0 <-------------> [Sep 3 12:02:23] --- (10 headers 0 lines) --- [Kpbx1*CLI> [Sep 3 12:02:23] Really destroying SIP dialog '1dc0b2b65c2a793a5ea477821e21b511@64.6.240.50' Method: OPTIONS [Kpbx1*CLI> [Sep 3 12:02:24] Reliably Transmitting (NAT) to 65.112.77.110:5060: OPTIONS sip:2726@65.112.77.110 SIP/2.0 Via: SIP/2.0/UDP 64.6.240.50:5060;branch=z9hG4bK5dd7f818;rport From: "Unknown" ;tag=as1f3de4e2 To: Contact: Call-ID: 485249154d71f580334274634e093ae5@64.6.240.50 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 03 Sep 2008 16:02:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Kpbx1*CLI> [Sep 3 12:02:24] Reliably Transmitting (NAT) to 65.112.77.109:5060: OPTIONS sip:2725@65.112.77.109 SIP/2.0 Via: SIP/2.0/UDP 64.6.240.50:5060;branch=z9hG4bK79f80628;rport From: "Unknown" ;tag=as7bef8d04 To: Contact: Call-ID: 510b8bda464db0513ae8209360dfdc92@64.6.240.50 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 03 Sep 2008 16:02:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Kpbx1*CLI> [Sep 3 12:02:24] <--- SIP read from 65.112.77.110:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 64.6.240.50:5060;branch=z9hG4bK5dd7f818;rport From: "Unknown" ;tag=as1f3de4e2 To: ;tag=FA54FFAA-3029C9D5 CSeq: 102 OPTIONS Call-ID: 485249154d71f580334274634e093ae5@64.6.240.50 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.3.0401 Content-Length: 0 <-------------> [Sep 3 12:02:24] --- (10 headers 0 lines) --- [Kpbx1*CLI> [Sep 3 12:02:24] Really destroying SIP dialog '485249154d71f580334274634e093ae5@64.6.240.50' Method: OPTIONS [Kpbx1*CLI> [Sep 3 12:02:24] <--- SIP read from 65.112.77.109:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 64.6.240.50:5060;branch=z9hG4bK79f80628;rport From: "Unknown" ;tag=as7bef8d04 To: ;tag=C68D6D9-9447D22E CSeq: 102 OPTIONS Call-ID: 510b8bda464db0513ae8209360dfdc92@64.6.240.50 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.3.0401 Content-Length: 0 <-------------> [Sep 3 12:02:24] --- (10 headers 0 lines) --- [Kpbx1*CLI> [Sep 3 12:02:24] Really destroying SIP dialog '510b8bda464db0513ae8209360dfdc92@64.6.240.50' Method: OPTIONS [Kpbx1*CLI> [Sep 3 12:02:24] Reliably Transmitting (NAT) to 65.112.77.107:5060: OPTIONS sip:2724@65.112.77.107 SIP/2.0 Via: SIP/2.0/UDP 64.6.240.50:5060;branch=z9hG4bK3b80d626;rport From: "Unknown" ;tag=as6dded8a8 To: Contact: Call-ID: 14cf7ff02b9661dd21830cef4608e079@64.6.240.50 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 03 Sep 2008 16:02:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Kpbx1*CLI> [Sep 3 12:02:24] <--- SIP read from 65.112.77.107:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 64.6.240.50:5060;branch=z9hG4bK3b80d626;rport From: "Unknown" ;tag=as6dded8a8 To: ;tag=F76C7C5D-E6660870 CSeq: 102 OPTIONS Call-ID: 14cf7ff02b9661dd21830cef4608e079@64.6.240.50 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.3.0401 Content-Length: 0 <-------------> [Sep 3 12:02:24] --- (10 headers 0 lines) --- [Kpbx1*CLI> [Sep 3 12:02:24] Really destroying SIP dialog '14cf7ff02b9661dd21830cef4608e079@64.6.240.50' Method: OPTIONS [Kpbx1*CLI> [Sep 3 12:02:24] Reliably Transmitting (NAT) to 65.112.77.115:5060: OPTIONS sip:2722@65.112.77.115 SIP/2.0 Via: SIP/2.0/UDP 64.6.240.50:5060;branch=z9hG4bK087ef8d8;rport From: "Unknown" ;tag=as36d56951 To: Contact: Call-ID: 73816ae01913e3054c0dcff623a59741@64.6.240.50 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 03 Sep 2008 16:02:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Kpbx1*CLI> [Sep 3 12:02:24] Reliably Transmitting (NAT) to 65.112.77.249:5060: OPTIONS sip:2721@65.112.77.249 SIP/2.0 Via: SIP/2.0/UDP 64.6.240.50:5060;branch=z9hG4bK7bc6c5f5;rport From: "Unknown" ;tag=as57182390 To: Contact: Call-ID: 035a2fdd5acda92a67ef61d4483253fc@64.6.240.50 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 03 Sep 2008 16:02:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Kpbx1*CLI> [Sep 3 12:02:24] <--- SIP read from 65.112.77.115:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 64.6.240.50:5060;branch=z9hG4bK087ef8d8;rport From: "Unknown" ;tag=as36d56951 To: ;tag=9F6EF9CC-3DA69B79 CSeq: 102 OPTIONS Call-ID: 73816ae01913e3054c0dcff623a59741@64.6.240.50 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.3.0401 Content-Length: 0 <-------------> [Sep 3 12:02:24] --- (10 headers 0 lines) --- [Kpbx1*CLI> [Sep 3 12:02:24] Really destroying SIP dialog '73816ae01913e3054c0dcff623a59741@64.6.240.50' Method: OPTIONS [Kpbx1*CLI> [Sep 3 12:02:24] <--- SIP read from 65.112.77.249:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 64.6.240.50:5060;branch=z9hG4bK7bc6c5f5;rport From: "Unknown" ;tag=as57182390 To: ;tag=4143C7F-3BCFBEAA CSeq: 102 OPTIONS Call-ID: 035a2fdd5acda92a67ef61d4483253fc@64.6.240.50 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.3.0401 Content-Length: 0 <-------------> [Sep 3 12:02:24] --- (10 headers 0 lines) --- [Kpbx1*CLI> [Sep 3 12:02:24] Really destroying SIP dialog '035a2fdd5acda92a67ef61d4483253fc@64.6.240.50' Method: OPTIONS [Kpbx1*CLI> [Sep 3 12:02:25] Reliably Transmitting (no NAT) to 65.112.77.100:5060: OPTIONS sip:2712@65.112.77.100 SIP/2.0 Via: SIP/2.0/UDP 64.6.240.50:5060;branch=z9hG4bK204b3724;rport From: "Unknown" ;tag=as2f7c7c30 To: Contact: Call-ID: 2ce6699773d717187083f2350fc1d718@64.6.240.50 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 03 Sep 2008 16:02:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Kpbx1*CLI> [Sep 3 12:02:25] <--- SIP read from 65.112.77.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 64.6.240.50:5060;branch=z9hG4bK204b3724;rport From: "Unknown" ;tag=as2f7c7c30 To: ;tag=6DDF6A1E-EF197701 CSeq: 102 OPTIONS Call-ID: 2ce6699773d717187083f2350fc1d718@64.6.240.50 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.3.0401 Content-Length: 0 <-------------> [Sep 3 12:02:25] --- (10 headers 0 lines) --- [Kpbx1*CLI> [Sep 3 12:02:25] Really destroying SIP dialog '2ce6699773d717187083f2350fc1d718@64.6.240.50' Method: OPTIONS [Kpbx1*CLI> [Sep 3 12:02:25] Reliably Transmitting (NAT) to 65.112.77.108:5060: OPTIONS sip:2711@65.112.77.108 SIP/2.0 Via: SIP/2.0/UDP 64.6.240.50:5060;branch=z9hG4bK4abad2a5;rport From: "Unknown" ;tag=as72a4b462 To: Contact: Call-ID: 565bb06777d6d36c1e36d5b71a2662bb@64.6.240.50 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 03 Sep 2008 16:02:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Kpbx1*CLI> [Sep 3 12:02:25] Reliably Transmitting (NAT) to 65.112.77.103:5060: OPTIONS sip:2710@65.112.77.103 SIP/2.0 Via: SIP/2.0/UDP 64.6.240.50:5060;branch=z9hG4bK1e6631cd;rport From: "Unknown" ;tag=as1ca4c778 To: Contact: Call-ID: 531a096732b20dc9348082fd225f5ba5@64.6.240.50 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 03 Sep 2008 16:02:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Kpbx1*CLI> [Sep 3 12:02:25] <--- SIP read from 65.112.77.103:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 64.6.240.50:5060;branch=z9hG4bK1e6631cd;rport From: "Unknown" ;tag=as1ca4c778 To: ;tag=B3603F6-A493BD91 CSeq: 102 OPTIONS Call-ID: 531a096732b20dc9348082fd225f5ba5@64.6.240.50 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.3.0401 Content-Length: 0 <-------------> [Sep 3 12:02:25] --- (10 headers 0 lines) --- [Kpbx1*CLI> [Sep 3 12:02:25] Really destroying SIP dialog '531a096732b20dc9348082fd225f5ba5@64.6.240.50' Method: OPTIONS [Kpbx1*CLI> [Sep 3 12:02:25] <--- SIP read from 65.112.77.108:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 64.6.240.50:5060;branch=z9hG4bK4abad2a5;rport From: "Unknown" ;tag=as72a4b462 To: ;tag=861C5BF0-333624BD CSeq: 102 OPTIONS Call-ID: 565bb06777d6d36c1e36d5b71a2662bb@64.6.240.50 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.3.0401 Content-Length: 0 <-------------> [Sep 3 12:02:25] --- (10 headers 0 lines) --- [Kpbx1*CLI> [Sep 3 12:02:25] Really destroying SIP dialog '565bb06777d6d36c1e36d5b71a2662bb@64.6.240.50' Method: OPTIONS [Kpbx1*CLI> [Sep 3 12:02:25] Reliably Transmitting (NAT) to 65.112.77.104:5060: OPTIONS sip:2709@65.112.77.104 SIP/2.0 Via: SIP/2.0/UDP 64.6.240.50:5060;branch=z9hG4bK7f2fa2c5;rport From: "Unknown" ;tag=as7359b35b To: Contact: Call-ID: 68074a202255ed336dd13afd24a6a44e@64.6.240.50 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 03 Sep 2008 16:02:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Kpbx1*CLI> [Sep 3 12:02:25] Reliably Transmitting (NAT) to 65.112.77.105:5060: OPTIONS sip:2708@65.112.77.105 SIP/2.0 Via: SIP/2.0/UDP 64.6.240.50:5060;branch=z9hG4bK768134a9;rport From: "Unknown" ;tag=as761284a3 To: Contact: Call-ID: 2368466b666dfb7d55fec6332d16e9c3@64.6.240.50 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 03 Sep 2008 16:02:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Kpbx1*CLI> [Sep 3 12:02:25] Reliably Transmitting (NAT) to 65.112.77.102:5060: OPTIONS sip:2705@65.112.77.102 SIP/2.0 Via: SIP/2.0/UDP 64.6.240.50:5060;branch=z9hG4bK5eb6b0e4;rport From: "Unknown" ;tag=as0a7d4509 To: Contact: Call-ID: 2f5e9cbf43b4ac02098e6429390ae059@64.6.240.50 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 03 Sep 2008 16:02:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Kpbx1*CLI> [Sep 3 12:02:25] <--- SIP read from 65.112.77.104:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 64.6.240.50:5060;branch=z9hG4bK7f2fa2c5;rport From: "Unknown" ;tag=as7359b35b To: ;tag=507110FF-C1F7CD8 CSeq: 102 OPTIONS Call-ID: 68074a202255ed336dd13afd24a6a44e@64.6.240.50 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 Content-Length: 0 <-------------> [Sep 3 12:02:25] --- (10 headers 0 lines) --- [Kpbx1*CLI> [Sep 3 12:02:25] Really destroying SIP dialog '68074a202255ed336dd13afd24a6a44e@64.6.240.50' Method: OPTIONS [Kpbx1*CLI> [Sep 3 12:02:25] <--- SIP read from 65.112.77.105:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 64.6.240.50:5060;branch=z9hG4bK768134a9;rport From: "Unknown" ;tag=as761284a3 To: ;tag=1A20FF8F-B78EDF2E CSeq: 102 OPTIONS Call-ID: 2368466b666dfb7d55fec6332d16e9c3@64.6.240.50 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.3.0401 Content-Length: 0 <-------------> [Sep 3 12:02:25] --- (10 headers 0 lines) --- [Kpbx1*CLI> [Sep 3 12:02:25] Really destroying SIP dialog '2368466b666dfb7d55fec6332d16e9c3@64.6.240.50' Method: OPTIONS [Kpbx1*CLI> [Sep 3 12:02:25] <--- SIP read from 65.112.77.102:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 64.6.240.50:5060;branch=z9hG4bK5eb6b0e4;rport From: "Unknown" ;tag=as0a7d4509 To: ;tag=523AF21C-54164D01 CSeq: 102 OPTIONS Call-ID: 2f5e9cbf43b4ac02098e6429390ae059@64.6.240.50 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.3.0401 Content-Length: 0 <-------------> [Sep 3 12:02:25] --- (10 headers 0 lines) --- [Kpbx1*CLI> [Sep 3 12:02:25] Really destroying SIP dialog '2f5e9cbf43b4ac02098e6429390ae059@64.6.240.50' Method: OPTIONS [Kpbx1*CLI> [Sep 3 12:02:26] Scheduling destruction of SIP dialog '1ac6c6825371e82004e4df45558f8aba@64.6.240.50' in 6656 ms (Method: INVITE) [Sep 3 12:02:26] set_destination: Parsing for address/port to send to [Sep 3 12:02:26] set_destination: set destination to 65.112.77.100, port 5060 [Sep 3 12:02:26] Reliably Transmitting (no NAT) to 65.112.77.100:5060: BYE sip:2712@65.112.77.100 SIP/2.0 Via: SIP/2.0/UDP 64.6.240.50:5060;branch=z9hG4bK5dd1542a;rport From: "Stefan Dimitrov" ;tag=as229109c5 To: ;tag=8F1E8693-401864E2 Call-ID: 1ac6c6825371e82004e4df45558f8aba@64.6.240.50 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Sep 3 12:02:26] == Spawn extension (macro-dial, s, 7) exited non-zero on 'SIP/2720-0933bac0' in macro 'dial' [Sep 3 12:02:26] == Spawn extension (macro-dial, s, 7) exited non-zero on 'SIP/2720-0933bac0' in macro 'exten-vm' [Sep 3 12:02:26] == Spawn extension (macro-dial, s, 7) exited non-zero on 'SIP/2720-0933bac0' [Sep 3 12:02:26] -- Executing [h@macro-dial:1] Macro("SIP/2720-0933bac0", "hangupcall") in new stack [Sep 3 12:02:26] -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/2720-0933bac0", "w") in new stack [Kpbx1*CLI> [Sep 3 12:02:26] -- Executing [s@macro-hangupcall:2] NoCDR("SIP/2720-0933bac0", "") in new stack [Kpbx1*CLI> [Sep 3 12:02:26] -- Executing [s@macro-hangupcall:3] GotoIf("SIP/2720-0933bac0", "1?skiprg") in new stack [Sep 3 12:02:26] -- Goto (macro-hangupcall,s,6) [Kpbx1*CLI> [Sep 3 12:02:26] -- Executing [s@macro-hangupcall:6] GotoIf("SIP/2720-0933bac0", "1?skipblkvm") in new stack [Kpbx1*CLI> [Sep 3 12:02:26] -- Goto (macro-hangupcall,s,9) [Kpbx1*CLI> [Sep 3 12:02:26] -- Executing [s@macro-hangupcall:9] GotoIf("SIP/2720-0933bac0", "1?theend") in new stack [Kpbx1*CLI> [Sep 3 12:02:26] -- Goto (macro-hangupcall,s,11) [Kpbx1*CLI> [Sep 3 12:02:26] -- Executing [s@macro-hangupcall:11] Hangup("SIP/2720-0933bac0", "") in new stack [Sep 3 12:02:26] == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/2720-0933bac0' in macro 'hangupcall' [Kpbx1*CLI> [Sep 3 12:02:26] == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/2720-0933bac0' [Sep 3 12:02:26] == End MixMonitor Recording SIP/2720-0933bac0 [Kpbx1*CLI> [Sep 3 12:02:26] <--- SIP read from 65.112.77.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 64.6.240.50:5060;branch=z9hG4bK5dd1542a;rport From: "Stefan Dimitrov" ;tag=as229109c5 To: ;tag=8F1E8693-401864E2 CSeq: 103 BYE Call-ID: 1ac6c6825371e82004e4df45558f8aba@64.6.240.50 Contact: User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.3.0401 Content-Length: 0 <-------------> [Kpbx1*CLI> [Sep 3 12:02:26] --- (9 headers 0 lines) --- [Kpbx1*CLI> [Sep 3 12:02:26] Really destroying SIP dialog '1ac6c6825371e82004e4df45558f8aba@64.6.240.50' Method: INVITE [Kpbx1*CLI> [Sep 3 12:02:26] Really destroying SIP dialog 'Y2JjMGI3MTU4M2YzZWY3ZDQ0M2U4NDc2MDAyYTBlZTI.' Method: INVITE [Kpbx1*CLI> quit Executing last minute cleanups [1;32m[[0;36mpbx1.myhostcenter.com[1;34m:[0;36m~[1;32m] [0;34m[1;34m([0mroot[0m[1;34m)[1;33m>[0mexit exit Script done on Wed Sep 3 12:02:29 2008