rendered paste bodyScript 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 <markster@digium.com>
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: <sip:2720@213.91.247.166:20228>
To: "Andy Mentges"<sip:2712@pbx1.myhostcenter.com>
From: <sip:2720@pbx1.myhostcenter.com>;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: <sip:2720@pbx1.myhostcenter.com>;tag=f650aa2c
To: "Andy Mentges"<sip:2712@pbx1.myhostcenter.com>;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"<sip:2712@pbx1.myhostcenter.com>;tag=as5321397e
From: <sip:2720@pbx1.myhostcenter.com>;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: <sip:2720@213.91.247.166:20228>
To: "Andy Mentges"<sip:2712@pbx1.myhostcenter.com>
From: <sip:2720@pbx1.myhostcenter.com>;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: <sip:2720@213.91.247.166:20228>
[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: <sip:2720@pbx1.myhostcenter.com>;tag=f650aa2c
To: "Andy Mentges"<sip:2712@pbx1.myhostcenter.com>
Call-ID: Y2JjMGI3MTU4M2YzZWY3ZDQ0M2U4NDc2MDAyYTBlZTI.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:2712@64.6.240.50>
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" <sip:2720@64.6.240.50>;tag=as229109c5
To: <sip:2712@65.112.77.100>
Contact: <sip:2720@64.6.240.50>
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: <sip:2720@pbx1.myhostcenter.com>;tag=f650aa2c
To: "Andy Mentges"<sip:2712@pbx1.myhostcenter.com>;tag=as28cef870
Call-ID: Y2JjMGI3MTU4M2YzZWY3ZDQ0M2U4NDc2MDAyYTBlZTI.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:2712@64.6.240.50>
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" <sip:2720@64.6.240.50>;tag=as229109c5
To: <sip:2712@65.112.77.100>;tag=8F1E8693-401864E2
CSeq: 102 INVITE
Call-ID: 1ac6c6825371e82004e4df45558f8aba@64.6.240.50
Contact: <sip:2712@65.112.77.100>
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" <sip:2720@64.6.240.50>;tag=as229109c5
To: <sip:2712@65.112.77.100>;tag=8F1E8693-401864E2
CSeq: 102 INVITE
Call-ID: 1ac6c6825371e82004e4df45558f8aba@64.6.240.50
Contact: <sip:2712@65.112.77.100>
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" <sip:2720@64.6.240.50>;tag=as229109c5
To: <sip:2712@65.112.77.100>;tag=8F1E8693-401864E2
CSeq: 102 INVITE
Call-ID: 1ac6c6825371e82004e4df45558f8aba@64.6.240.50
Contact: <sip:2712@65.112.77.100>
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: <sip:2712@65.112.77.100>
[Sep 3 12:02:06] set_destination: Parsing <sip:2712@65.112.77.100> 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" <sip:2720@64.6.240.50>;tag=as229109c5
To: <sip:2712@65.112.77.100>;tag=8F1E8693-401864E2
Contact: <sip:2720@64.6.240.50>
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: <sip:2720@pbx1.myhostcenter.com>;tag=f650aa2c
To: "Andy Mentges"<sip:2712@pbx1.myhostcenter.com>;tag=as28cef870
Call-ID: Y2JjMGI3MTU4M2YzZWY3ZDQ0M2U4NDc2MDAyYTBlZTI.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:2712@64.6.240.50>
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: <sip:2720@pbx1.myhostcenter.com>;tag=f650aa2c
To: "Andy Mentges"<sip:2712@pbx1.myhostcenter.com>;tag=as28cef870
Call-ID: Y2JjMGI3MTU4M2YzZWY3ZDQ0M2U4NDc2MDAyYTBlZTI.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:2712@64.6.240.50>
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: <sip:2720@pbx1.myhostcenter.com>;tag=f650aa2c
To: "Andy Mentges"<sip:2712@pbx1.myhostcenter.com>;tag=as28cef870
Call-ID: Y2JjMGI3MTU4M2YzZWY3ZDQ0M2U4NDc2MDAyYTBlZTI.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:2712@64.6.240.50>
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: <sip:2720@pbx1.myhostcenter.com>;tag=f650aa2c
To: "Andy Mentges"<sip:2712@pbx1.myhostcenter.com>;tag=as28cef870
Call-ID: Y2JjMGI3MTU4M2YzZWY3ZDQ0M2U4NDc2MDAyYTBlZTI.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:2712@64.6.240.50>
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: <sip:2720@pbx1.myhostcenter.com>;tag=f650aa2c
To: "Andy Mentges"<sip:2712@pbx1.myhostcenter.com>;tag=as28cef870
Call-ID: Y2JjMGI3MTU4M2YzZWY3ZDQ0M2U4NDc2MDAyYTBlZTI.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:2712@64.6.240.50>
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: <sip:2720@pbx1.myhostcenter.com>;tag=f650aa2c
To: "Andy Mentges"<sip:2712@pbx1.myhostcenter.com>;tag=as28cef870
Call-ID: Y2JjMGI3MTU4M2YzZWY3ZDQ0M2U4NDc2MDAyYTBlZTI.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:2712@64.6.240.50>
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" <sip:Unknown@64.6.240.50>;tag=as25f7bbff
To: <sip:2731@64.6.243.44>
Contact: <sip:Unknown@64.6.240.50>
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" <sip:Unknown@64.6.240.50>;tag=as25f7bbff
To: <sip:2731@64.6.243.44>;tag=5CEB3FB2-8B02E449
CSeq: 102 OPTIONS
Call-ID: 4407d6ac449746741800de764813b861@64.6.240.50
Contact: <sip:2731@64.6.243.44>
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: <sip:2720@pbx1.myhostcenter.com>;tag=f650aa2c
To: "Andy Mentges"<sip:2712@pbx1.myhostcenter.com>;tag=as28cef870
Call-ID: Y2JjMGI3MTU4M2YzZWY3ZDQ0M2U4NDc2MDAyYTBlZTI.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:2712@64.6.240.50>
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" <sip:Unknown@64.6.240.50>;tag=as2485d389
To: <sip:2798@192.168.0.50>
Contact: <sip:Unknown@64.6.240.50>
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" <sip:Unknown@64.6.240.50>;tag=as5797c74b
To: <sip:2799@192.168.2.10>
Contact: <sip:Unknown@64.6.240.50>
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" <sip:Unknown@64.6.240.50>;tag=as2485d389
To: <sip:2798@192.168.0.50>;tag=9D0368E9-6FF366EC
CSeq: 102 OPTIONS
Call-ID: 3ba09719030a657655574be242745e35@64.6.240.50
Contact: <sip:2798@192.168.0.50>
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" <sip:Unknown@64.6.240.50>;tag=as5797c74b
To: <sip:2799@192.168.2.10>;tag=6FAACDD0-13A77657
CSeq: 102 OPTIONS
Call-ID: 04e43c651448b96b3da3c55e6cc266fe@64.6.240.50
Contact: <sip:2799@192.168.2.10>
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" <sip:Unknown@64.6.240.50>;tag=as6042f228
To: <sip:2723@192.168.1.159>
Contact: <sip:Unknown@64.6.240.50>
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" <sip:Unknown@64.6.240.50>;tag=as6042f228
To: <sip:2723@192.168.1.159>;tag=57E68532-64E9F3C1
CSeq: 102 OPTIONS
Call-ID: 1dc0b2b65c2a793a5ea477821e21b511@64.6.240.50
Contact: <sip:2723@192.168.1.159>
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" <sip:Unknown@64.6.240.50>;tag=as1f3de4e2
To: <sip:2726@65.112.77.110>
Contact: <sip:Unknown@64.6.240.50>
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" <sip:Unknown@64.6.240.50>;tag=as7bef8d04
To: <sip:2725@65.112.77.109>
Contact: <sip:Unknown@64.6.240.50>
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" <sip:Unknown@64.6.240.50>;tag=as1f3de4e2
To: <sip:2726@65.112.77.110>;tag=FA54FFAA-3029C9D5
CSeq: 102 OPTIONS
Call-ID: 485249154d71f580334274634e093ae5@64.6.240.50
Contact: <sip:2726@65.112.77.110>
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" <sip:Unknown@64.6.240.50>;tag=as7bef8d04
To: <sip:2725@65.112.77.109>;tag=C68D6D9-9447D22E
CSeq: 102 OPTIONS
Call-ID: 510b8bda464db0513ae8209360dfdc92@64.6.240.50
Contact: <sip:2725@65.112.77.109>
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" <sip:Unknown@64.6.240.50>;tag=as6dded8a8
To: <sip:2724@65.112.77.107>
Contact: <sip:Unknown@64.6.240.50>
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" <sip:Unknown@64.6.240.50>;tag=as6dded8a8
To: <sip:2724@65.112.77.107>;tag=F76C7C5D-E6660870
CSeq: 102 OPTIONS
Call-ID: 14cf7ff02b9661dd21830cef4608e079@64.6.240.50
Contact: <sip:2724@65.112.77.107>
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" <sip:Unknown@64.6.240.50>;tag=as36d56951
To: <sip:2722@65.112.77.115>
Contact: <sip:Unknown@64.6.240.50>
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" <sip:Unknown@64.6.240.50>;tag=as57182390
To: <sip:2721@65.112.77.249>
Contact: <sip:Unknown@64.6.240.50>
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" <sip:Unknown@64.6.240.50>;tag=as36d56951
To: <sip:2722@65.112.77.115>;tag=9F6EF9CC-3DA69B79
CSeq: 102 OPTIONS
Call-ID: 73816ae01913e3054c0dcff623a59741@64.6.240.50
Contact: <sip:2722@65.112.77.115>
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" <sip:Unknown@64.6.240.50>;tag=as57182390
To: <sip:2721@65.112.77.249>;tag=4143C7F-3BCFBEAA
CSeq: 102 OPTIONS
Call-ID: 035a2fdd5acda92a67ef61d4483253fc@64.6.240.50
Contact: <sip:2721@65.112.77.249>
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" <sip:Unknown@64.6.240.50>;tag=as2f7c7c30
To: <sip:2712@65.112.77.100>
Contact: <sip:Unknown@64.6.240.50>
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" <sip:Unknown@64.6.240.50>;tag=as2f7c7c30
To: <sip:2712@65.112.77.100>;tag=6DDF6A1E-EF197701
CSeq: 102 OPTIONS
Call-ID: 2ce6699773d717187083f2350fc1d718@64.6.240.50
Contact: <sip:2712@65.112.77.100>
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" <sip:Unknown@64.6.240.50>;tag=as72a4b462
To: <sip:2711@65.112.77.108>
Contact: <sip:Unknown@64.6.240.50>
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" <sip:Unknown@64.6.240.50>;tag=as1ca4c778
To: <sip:2710@65.112.77.103>
Contact: <sip:Unknown@64.6.240.50>
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" <sip:Unknown@64.6.240.50>;tag=as1ca4c778
To: <sip:2710@65.112.77.103>;tag=B3603F6-A493BD91
CSeq: 102 OPTIONS
Call-ID: 531a096732b20dc9348082fd225f5ba5@64.6.240.50
Contact: <sip:2710@65.112.77.103>
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" <sip:Unknown@64.6.240.50>;tag=as72a4b462
To: <sip:2711@65.112.77.108>;tag=861C5BF0-333624BD
CSeq: 102 OPTIONS
Call-ID: 565bb06777d6d36c1e36d5b71a2662bb@64.6.240.50
Contact: <sip:2711@65.112.77.108>
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" <sip:Unknown@64.6.240.50>;tag=as7359b35b
To: <sip:2709@65.112.77.104>
Contact: <sip:Unknown@64.6.240.50>
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" <sip:Unknown@64.6.240.50>;tag=as761284a3
To: <sip:2708@65.112.77.105>
Contact: <sip:Unknown@64.6.240.50>
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" <sip:Unknown@64.6.240.50>;tag=as0a7d4509
To: <sip:2705@65.112.77.102>
Contact: <sip:Unknown@64.6.240.50>
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" <sip:Unknown@64.6.240.50>;tag=as7359b35b
To: <sip:2709@65.112.77.104>;tag=507110FF-C1F7CD8
CSeq: 102 OPTIONS
Call-ID: 68074a202255ed336dd13afd24a6a44e@64.6.240.50
Contact: <sip:2709@65.112.77.104>
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" <sip:Unknown@64.6.240.50>;tag=as761284a3
To: <sip:2708@65.112.77.105>;tag=1A20FF8F-B78EDF2E
CSeq: 102 OPTIONS
Call-ID: 2368466b666dfb7d55fec6332d16e9c3@64.6.240.50
Contact: <sip:2708@65.112.77.105>
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" <sip:Unknown@64.6.240.50>;tag=as0a7d4509
To: <sip:2705@65.112.77.102>;tag=523AF21C-54164D01
CSeq: 102 OPTIONS
Call-ID: 2f5e9cbf43b4ac02098e6429390ae059@64.6.240.50
Contact: <sip:2705@65.112.77.102>
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 <sip:2712@65.112.77.100> 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" <sip:2720@64.6.240.50>;tag=as229109c5
To: <sip:2712@65.112.77.100>;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" <sip:2720@64.6.240.50>;tag=as229109c5
To: <sip:2712@65.112.77.100>;tag=8F1E8693-401864E2
CSeq: 103 BYE
Call-ID: 1ac6c6825371e82004e4df45558f8aba@64.6.240.50
Contact: <sip:2712@65.112.77.100>
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