colopbx2*CLI> 
<--- SIP read from 192.168.1.13:48182 --->
INVITE sip:2002@192.168.1.64 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.13:48182;branch=z9hG4bK-d87543-2854e61fd12be50d-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:190@192.168.1.13:48182>
To: "2002"<sip:2002@192.168.1.64>
From: "190"<sip:190@192.168.1.64>;tag=3e3ccd75
Call-ID: ODRlOTJkYzIzNjY1NDU3ZTAwMmU4ZTk0ZGM1N2YzNjA.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1006e stamp 34025
Content-Length: 325

v=0
o=- 3 2 IN IP4 192.168.1.13
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.1.13
t=0 0
m=audio 58828 RTP/AVP 107 119 0 98 8 3 101
a=alt:1 1 : MUEK5BpJ tuXNjMtP 192.168.1.13 58828
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
--- (12 headers 13 lines) ---
Sending to 192.168.1.13 : 48182 (NAT)
Using INVITE request as basis request - ODRlOTJkYzIzNjY1NDU3ZTAwMmU4ZTk0ZGM1N2YzNjA.
colopbx2*CLI> 
<--- Reliably Transmitting (no NAT) to 192.168.1.13:48182 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.13:48182;branch=z9hG4bK-d87543-2854e61fd12be50d-1--d87543-;received=192.168.1.13;rport=48182
From: "190"<sip:190@192.168.1.64>;tag=3e3ccd75
To: "2002"<sip:2002@192.168.1.64>;tag=as4ae13ac9
Call-ID: ODRlOTJkYzIzNjY1NDU3ZTAwMmU4ZTk0ZGM1N2YzNjA.
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="0cbb6ca6"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'ODRlOTJkYzIzNjY1NDU3ZTAwMmU4ZTk0ZGM1N2YzNjA.' in 32000 ms (Method: INVITE)
Found user '190'
colopbx2*CLI> 
<--- SIP read from 192.168.1.13:48182 --->
ACK sip:2002@192.168.1.64 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.13:48182;branch=z9hG4bK-d87543-2854e61fd12be50d-1--d87543-;rport
To: "2002"<sip:2002@192.168.1.64>;tag=as4ae13ac9
From: "190"<sip:190@192.168.1.64>;tag=3e3ccd75
Call-ID: ODRlOTJkYzIzNjY1NDU3ZTAwMmU4ZTk0ZGM1N2YzNjA.
CSeq: 1 ACK
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
colopbx2*CLI> 
<--- SIP read from 192.168.1.13:48182 --->
INVITE sip:2002@192.168.1.64 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.13:48182;branch=z9hG4bK-d87543-323bd563156ef31a-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:190@192.168.1.13:48182>
To: "2002"<sip:2002@192.168.1.64>
From: "190"<sip:190@192.168.1.64>;tag=3e3ccd75
Call-ID: ODRlOTJkYzIzNjY1NDU3ZTAwMmU4ZTk0ZGM1N2YzNjA.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest username="190",realm="asterisk",nonce="0cbb6ca6",uri="sip:2002@192.168.1.64",response="d6673fb5213f713ee371452749d77dd1",algorithm=MD5
User-Agent: X-Lite release 1006e stamp 34025
Content-Length: 325

v=0
o=- 3 2 IN IP4 192.168.1.13
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.1.13
t=0 0
m=audio 58828 RTP/AVP 107 119 0 98 8 3 101
a=alt:1 1 : MUEK5BpJ tuXNjMtP 192.168.1.13 58828
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
--- (13 headers 13 lines) ---
Sending to 192.168.1.13 : 48182 (NAT)
Using INVITE request as basis request - ODRlOTJkYzIzNjY1NDU3ZTAwMmU4ZTk0ZGM1N2YzNjA.
Found user '190'
Found RTP audio format 107
Found RTP audio format 119
Found RTP audio format 0
Found RTP audio format 98
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.13:58828
Found description format BV32 for ID 107
Found description format BV32-FEC for ID 119
Found description format iLBC for ID 98
Found description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.13:58828
Looking for 2002 in from-internal (domain 192.168.1.64)
list_route: hop: <sip:190@192.168.1.13:48182>
colopbx2*CLI> 
<--- Transmitting (no NAT) to 192.168.1.13:48182 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.13:48182;branch=z9hG4bK-d87543-323bd563156ef31a-1--d87543-;received=192.168.1.13;rport=48182
From: "190"<sip:190@192.168.1.64>;tag=3e3ccd75
To: "2002"<sip:2002@192.168.1.64>
Call-ID: ODRlOTJkYzIzNjY1NDU3ZTAwMmU4ZTk0ZGM1N2YzNjA.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:2002@192.168.1.64>
Content-Length: 0


<------------>
    -- Executing [2002@from-internal:1] Macro("SIP/190-b7c1a128", "user-callerid|") in new stack
    -- Executing [s@macro-user-callerid:1] NoOp("SIP/190-b7c1a128", "user-callerid: device 190") in new stack
    -- Executing [s@macro-user-callerid:2] Set("SIP/190-b7c1a128", "AMPUSER=190") in new stack
    -- Executing [s@macro-user-callerid:3] GotoIf("SIP/190-b7c1a128", "0?report") in new stack
    -- Executing [s@macro-user-callerid:4] GotoIf("SIP/190-b7c1a128", "0?start") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/190-b7c1a128", "REALCALLERIDNUM=190") in new stack
    -- Executing [s@macro-user-callerid:6] NoOp("SIP/190-b7c1a128", "REALCALLERIDNUM is 190") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/190-b7c1a128", "AMPUSER=190") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/190-b7c1a128", "AMPUSERCIDNAME=Test1") in new stack
    -- Executing [s@macro-user-callerid:9] GotoIf("SIP/190-b7c1a128", "0?report") in new stack
    -- Executing [s@macro-user-callerid:10] Set("SIP/190-b7c1a128", "AMPUSERCID=190") in new stack
    -- Executing [s@macro-user-callerid:11] Set("SIP/190-b7c1a128", "CALLERID(all)="Test1" <190>") in new stack
    -- Executing [s@macro-user-callerid:12] Set("SIP/190-b7c1a128", "REALCALLERIDNUM=190") in new stack
    -- Executing [s@macro-user-callerid:13] NoOp("SIP/190-b7c1a128", "TTL:  ARG1: ") in new stack
    -- Executing [s@macro-user-callerid:14] GotoIf("SIP/190-b7c1a128", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:15] Set("SIP/190-b7c1a128", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:16] GotoIf("SIP/190-b7c1a128", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,23)
    -- Executing [s@macro-user-callerid:23] NoOp("SIP/190-b7c1a128", "Using CallerID "Test1" <190>") in new stack
    -- Executing [2002@from-internal:2] Answer("SIP/190-b7c1a128", "") in new stack
Audio is at 192.168.1.64 port 14516
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
colopbx2*CLI> 
<--- Reliably Transmitting (no NAT) to 192.168.1.13:48182 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.13:48182;branch=z9hG4bK-d87543-323bd563156ef31a-1--d87543-;received=192.168.1.13;rport=48182
From: "190"<sip:190@192.168.1.64>;tag=3e3ccd75
To: "2002"<sip:2002@192.168.1.64>;tag=as644f9306
Call-ID: ODRlOTJkYzIzNjY1NDU3ZTAwMmU4ZTk0ZGM1N2YzNjA.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:2002@192.168.1.64>
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 30199 30199 IN IP4 192.168.1.64
s=session
c=IN IP4 192.168.1.64
t=0 0
m=audio 14516 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
    -- Executing [2002@from-internal:3] Set("SIP/190-b7c1a128", "__BLKVM_OVERRIDE=BLKVM/2002/SIP/190-b7c1a128") in new stack
    -- Executing [2002@from-internal:4] Set("SIP/190-b7c1a128", "__BLKVM_BASE=2002") in new stack
    -- Executing [2002@from-internal:5] Set("SIP/190-b7c1a128", "DB(BLKVM/2002/SIP/190-b7c1a128)=TRUE") in new stack
    -- Executing [2002@from-internal:6] Set("SIP/190-b7c1a128", "_DIAL_OPTIONS=trwM(auto-blkvm)") in new stack
    -- Executing [2002@from-internal:7] Set("SIP/190-b7c1a128", "__NODEST=2002") in new stack
    -- Executing [2002@from-internal:8] Set("SIP/190-b7c1a128", "MONITOR_FILENAME=/var/spool/asterisk/monitor/q2002-20070905-153634-1189031794.94") in new stack
    -- Executing [2002@from-internal:9] Set("SIP/190-b7c1a128", "__MOHCLASS=custom") in new stack
    -- Executing [2002@from-internal:10] Queue("SIP/190-b7c1a128", "2002|t||") in new stack
    -- Started music on hold, class 'custom', on channel 'SIP/190-b7c1a128'
    -- Executing [192@from-internal:1] Macro("Local/192@from-internal-3eb9,2", "exten-vm|192|192") in new stack
    -- Executing [s@macro-exten-vm:1] Macro("Local/192@from-internal-3eb9,2", "user-callerid") in new stack
    -- Executing [s@macro-user-callerid:1] NoOp("Local/192@from-internal-3eb9,2", "user-callerid: Test1 190") in new stack
    -- Executing [s@macro-user-callerid:2] Set("Local/192@from-internal-3eb9,2", "AMPUSER=190") in new stack
    -- Executing [s@macro-user-callerid:3] GotoIf("Local/192@from-internal-3eb9,2", "1?report") in new stack
    -- Goto (macro-user-callerid,s,13)
    -- Executing [s@macro-user-callerid:13] NoOp("Local/192@from-internal-3eb9,2", "TTL: 64 ARG1: 192") in new stack
    -- Executing [s@macro-user-callerid:14] GotoIf("Local/192@from-internal-3eb9,2", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:15] Set("Local/192@from-internal-3eb9,2", "__TTL=63") in new stack
    -- Executing [s@macro-user-callerid:16] GotoIf("Local/192@from-internal-3eb9,2", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,23)
    -- Executing [s@macro-user-callerid:23] NoOp("Local/192@from-internal-3eb9,2", "Using CallerID "Test1" <190>") in new stack
    -- Executing [s@macro-exten-vm:2] Set("Local/192@from-internal-3eb9,2", "FROMCONTEXT=exten-vm") in new stack
    -- Executing [s@macro-exten-vm:3] Set("Local/192@from-internal-3eb9,2", "VMBOX=192") in new stack
    -- Executing [s@macro-exten-vm:4] Set("Local/192@from-internal-3eb9,2", "EXTTOCALL=192") in new stack
    -- Executing [s@macro-exten-vm:5] Set("Local/192@from-internal-3eb9,2", "CFUEXT=") in new stack
    -- Executing [s@macro-exten-vm:6] Set("Local/192@from-internal-3eb9,2", "CFBEXT=") in new stack
    -- Executing [s@macro-exten-vm:7] Set("Local/192@from-internal-3eb9,2", "RT=20") in new stack
    -- Executing [s@macro-exten-vm:8] Macro("Local/192@from-internal-3eb9,2", "record-enable|192|IN") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("Local/192@from-internal-3eb9,2", "0?2:4") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] DeadAGI("Local/192@from-internal-3eb9,2", "recordingcheck|20070905-153634|1189031794.96") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
  recordingcheck|20070905-153634|1189031794.96: Inbound recording not enabled
    -- AGI Script recordingcheck completed, returning 0
    -- Executing [s@macro-record-enable:5] NoOp("Local/192@from-internal-3eb9,2", "No recording needed") in new stack
    -- Executing [s@macro-exten-vm:9] Macro("Local/192@from-internal-3eb9,2", "dial|20|trwM(auto-blkvm)|192") in new stack
    -- Executing [s@macro-dial:1] GotoIf("Local/192@from-internal-3eb9,2", "0?dial") in new stack
    -- Executing [s@macro-dial:2] SetMusicOnHold("Local/192@from-internal-3eb9,2", "custom") in new stack
    -- Executing [s@macro-dial:3] AGI("Local/192@from-internal-3eb9,2", "dialparties.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
  dialparties.agi: Starting New Dialparties.agi
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'admin' logged on from 127.0.0.1
  dialparties.agi: Caller ID name is 'Test1' number is '190'
  dialparties.agi: Methodology of ring is  'none'
    --  dialparties.agi: Added extension 192 to extension map
    --  dialparties.agi: Extension 192 cf is disabled
    --  dialparties.agi: Extension 192 do not disturb is disabled
    --  dialparties.agi: dbset CALLTRACE/192 to 190
  == Manager 'admin' logged off from 127.0.0.1
    -- AGI Script dialparties.agi completed, returning 0
    -- Executing [s@macro-dial:10] Dial("Local/192@from-internal-3eb9,2", "SIP/192|20|trwM(auto-blkvm)") in new stack
Audio is at 192.168.1.64 port 10834
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.13:5060:
INVITE sip:192@192.168.1.13:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.64:5060;branch=z9hG4bK052a11ae;rport
From: "Test1" <sip:190@192.168.1.64>;tag=as4ac3ca32
To: <sip:192@192.168.1.13:5060>
Contact: <sip:190@192.168.1.64>
Call-ID: 2f771e5a396f08cb4569a064246c8ee7@192.168.1.64
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 05 Sep 2007 22:36:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 30199 30199 IN IP4 192.168.1.64
s=session
c=IN IP4 192.168.1.64
t=0 0
m=audio 10834 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 192
    -- Local/192@from-internal-3eb9,1 is ringing
colopbx2*CLI> 
<--- SIP read from 192.168.1.13:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.64:5060;rport=5060;received=192.168.1.64;branch=z9hG4bK052a11ae
Content-Length: 0
Call-ID: 2f771e5a396f08cb4569a064246c8ee7@192.168.1.64
CSeq: 102 INVITE
From: "Test1"<sip:190@192.168.1.64>;tag=as4ac3ca32
Server: SJphone/1.60.289a (SJ Labs)
To: "unknown"<sip:192@192.168.1.13:5060>;tag=45926400012549


<------------->
--- (8 headers 0 lines) ---
colopbx2*CLI> 
<--- SIP read from 192.168.1.13:48182 --->
ACK sip:2002@192.168.1.64 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.13:48182;branch=z9hG4bK-d87543-07091728db0b5e61-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:190@192.168.1.13:48182>
To: "2002"<sip:2002@192.168.1.64>;tag=as644f9306
From: "190"<sip:190@192.168.1.64>;tag=3e3ccd75
Call-ID: ODRlOTJkYzIzNjY1NDU3ZTAwMmU4ZTk0ZGM1N2YzNjA.
CSeq: 2 ACK
Proxy-Authorization: Digest username="190",realm="asterisk",nonce="0cbb6ca6",uri="sip:2002@192.168.1.64",response="d6673fb5213f713ee371452749d77dd1",algorithm=MD5
User-Agent: X-Lite release 1006e stamp 34025
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
colopbx2*CLI> 
<--- SIP read from 192.168.1.13:5060 --->
OPTIONS sip:192.168.1.64 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.13;rport;branch=z9hG4bKc0a8010d0000001046df2fc200002fcc00000057
Content-Length: 0
Call-ID: 51B44580-9265-470D-936A-EDB31EA5BA16@192.168.1.13
CSeq: 22 OPTIONS
From: <sip:192@192.168.1.64>;tag=45926417129582
Max-Forwards: 70
To: <sip:192.168.1.64>


<------------->
--- (8 headers 0 lines) ---
Looking for s in from-sip-external (domain 192.168.1.64)
colopbx2*CLI> 
<--- Transmitting (no NAT) to 192.168.1.13:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.13;branch=z9hG4bKc0a8010d0000001046df2fc200002fcc00000057;received=192.168.1.13;rport=5060
From: <sip:192@192.168.1.64>;tag=45926417129582
To: <sip:192.168.1.64>;tag=as75c79d09
Call-ID: 51B44580-9265-470D-936A-EDB31EA5BA16@192.168.1.13
CSeq: 22 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:192.168.1.64>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '51B44580-9265-470D-936A-EDB31EA5BA16@192.168.1.13' in 32000 ms (Method: OPTIONS)
colopbx2*CLI> 
<--- SIP read from 192.168.1.13:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.64:5060;rport=5060;received=192.168.1.64;branch=z9hG4bK052a11ae
Content-Length: 0
Contact: <sip:192@192.168.1.13:5060>
Call-ID: 2f771e5a396f08cb4569a064246c8ee7@192.168.1.64
CSeq: 102 INVITE
From: "Test1"<sip:190@192.168.1.64>;tag=as4ac3ca32
Server: SJphone/1.60.289a (SJ Labs)
To: "unknown"<sip:192@192.168.1.13:5060>;tag=45926400012549


<------------->
--- (9 headers 0 lines) ---
    -- SIP/192-08b2ced0 is ringing



"OPTION WAS PRESSED"


Scheduling destruction of SIP dialog '2f771e5a396f08cb4569a064246c8ee7@192.168.1.64' in 6400 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 192.168.1.13:5060:
CANCEL sip:192@192.168.1.13:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.64:5060;branch=z9hG4bK052a11ae;rport
From: "Test1" <sip:190@192.168.1.64>;tag=as4ac3ca32
To: <sip:192@192.168.1.13:5060>
Call-ID: 2f771e5a396f08cb4569a064246c8ee7@192.168.1.64
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Scheduling destruction of SIP dialog '2f771e5a396f08cb4569a064246c8ee7@192.168.1.64' in 6400 ms (Method: INVITE)
  == Spawn extension (macro-dial, s, 10) exited non-zero on 'Local/192@from-internal-3eb9,2' in macro 'dial'
  == Spawn extension (macro-dial, s, 10) exited non-zero on 'Local/192@from-internal-3eb9,2' in macro 'exten-vm'
  == Spawn extension (macro-dial, s, 10) exited non-zero on 'Local/192@from-internal-3eb9,2'
    -- Executing [h@macro-dial:1] Macro("Local/192@from-internal-3eb9,2", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] ResetCDR("Local/192@from-internal-3eb9,2", "w") in new stack
    -- Executing [s@macro-hangupcall:2] NoCDR("Local/192@from-internal-3eb9,2", "") in new stack
    -- Executing [s@macro-hangupcall:3] GotoIf("Local/192@from-internal-3eb9,2", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,6)
    -- Executing [s@macro-hangupcall:6] GotoIf("Local/192@from-internal-3eb9,2", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] GotoIf("Local/192@from-internal-3eb9,2", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,11)
    -- Executing [s@macro-hangupcall:11] Hangup("Local/192@from-internal-3eb9,2", "") in new stack
  == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'Local/192@from-internal-3eb9,2' in macro 'hangupcall'
  == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'Local/192@from-internal-3eb9,2'
colopbx2*CLI> 
<--- SIP read from 192.168.1.13:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.64:5060;rport=5060;received=192.168.1.64;branch=z9hG4bK052a11ae
Content-Length: 0
Call-ID: 2f771e5a396f08cb4569a064246c8ee7@192.168.1.64
CSeq: 102 CANCEL
From: "Test1"<sip:190@192.168.1.64>;tag=as4ac3ca32
Server: SJphone/1.60.289a (SJ Labs)
To: "unknown"<sip:192@192.168.1.13:5060>;tag=45926400012549


<------------->
--- (8 headers 0 lines) ---
colopbx2*CLI> 
<--- SIP read from 192.168.1.13:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.64:5060;rport=5060;received=192.168.1.64;branch=z9hG4bK052a11ae
Content-Length: 0
Call-ID: 2f771e5a396f08cb4569a064246c8ee7@192.168.1.64
CSeq: 102 INVITE
From: "Test1"<sip:190@192.168.1.64>;tag=as4ac3ca32
Server: SJphone/1.60.289a (SJ Labs)
To: "unknown"<sip:192@192.168.1.13:5060>;tag=45926400012549


<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 192.168.1.13:5060:
ACK sip:192@192.168.1.13:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.64:5060;branch=z9hG4bK052a11ae;rport
From: "Test1" <sip:190@192.168.1.64>;tag=as4ac3ca32
To: <sip:192@192.168.1.13:5060>;tag=45926400012549
Contact: <sip:190@192.168.1.64>
Call-ID: 2f771e5a396f08cb4569a064246c8ee7@192.168.1.64
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Really destroying SIP dialog '2f771e5a396f08cb4569a064246c8ee7@192.168.1.64' Method: INVITE


"OPTION WAS PRESSED"


    -- Stopped music on hold on SIP/190-b7c1a128
    -- Executing [5@ivr-9:1] DBdel("SIP/190-b7c1a128", "BLKVM/2002/SIP/190-b7c1a128") in new stack
    -- DBdel: family=BLKVM, key=2002/SIP/190-b7c1a128
    -- Executing [5@ivr-9:2] Set("SIP/190-b7c1a128", "__NODEST=") in new stack
    -- Executing [5@ivr-9:3] Goto("SIP/190-b7c1a128", "app-announcement-17|s|1") in new stack
    -- Goto (app-announcement-17,s,1)
    -- Executing [s@app-announcement-17:1] GotoIf("SIP/190-b7c1a128", "1?begin") in new stack
    -- Goto (app-announcement-17,s,4)
    -- Executing [s@app-announcement-17:4] NoOp("SIP/190-b7c1a128", "Playing announcement VMPrompt") in new stack
    -- Executing [s@app-announcement-17:5] Playback("SIP/190-b7c1a128", "custom/VMPrompt|noanswer") in new stack
    -- <SIP/190-b7c1a128> Playing 'custom/VMPrompt' (language 'en')
