rendered paste body
Asterisk 1.2.23, Copyright (C) 1999 - 2007 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type '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 'show license' for details.
=========================================================================
Connected to Asterisk 1.2.23 currently running on asterisk1 (pid = 3653)
asterisk1*CLI>
12 headers, 0 lines
[Kasterisk1*CLI>
Reliably Transmitting (NAT) to 64.192.196.12:5060:
OPTIONS sip:64.192.196.12 SIP/2.0
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK74cf4c47;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as4af11bbe
To: <sip:64.192.196.12>
Contact: <sip:Unknown@208.75.208.90>
Call-ID: 6f5639dc69f27a181f2e435d051fd97c@208.75.208.90
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 09 Nov 2007 01:53:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
[Kasterisk1*CLI>
<-- SIP read from 64.192.196.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK74cf4c47;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as4af11bbe
To: <sip:64.192.196.12>;tag=29FDE034-696
Date: Sat, 09 Mar 2002 03:41:43 GMT
Call-ID: 6f5639dc69f27a181f2e435d051fd97c@208.75.208.90
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,resource-priority,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 451
v=0
o=CiscoSystemsSIP-GW-UserAgent 5120 7001 IN IP4 64.192.196.12
s=SIP Call
c=IN IP4 64.192.196.12
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 15 3
c=IN IP4 64.192.196.12
m=image 0 udptl t38
c=IN IP4 64.192.196.12
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
[Kasterisk1*CLI>
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
--- (14 headers 18 lines) ---
[Kasterisk1*CLI>
Destroying call '6f5639dc69f27a181f2e435d051fd97c@208.75.208.90'
[Kasterisk1*CLI>
12 headers, 0 lines
Reliably Transmitting (NAT) to 64.192.196.12:5060:
OPTIONS sip:64.192.196.12 SIP/2.0
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK7ba55399;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as6ea75e75
To: <sip:64.192.196.12>
Contact: <sip:Unknown@208.75.208.90>
Call-ID: 79f1c5c80cfc56d5759ed08b150075a0@208.75.208.90
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 09 Nov 2007 01:53:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
[Kasterisk1*CLI>
<-- SIP read from 64.192.196.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK7ba55399;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as6ea75e75
To: <sip:64.192.196.12>;tag=29FDE06C-46F
Date: Sat, 09 Mar 2002 03:41:43 GMT
Call-ID: 79f1c5c80cfc56d5759ed08b150075a0@208.75.208.90
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,resource-priority,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 451
v=0
o=CiscoSystemsSIP-GW-UserAgent 1612 6599 IN IP4 64.192.196.12
s=SIP Call
c=IN IP4 64.192.196.12
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 15 3
c=IN IP4 64.192.196.12
m=image 0 udptl t38
c=IN IP4 64.192.196.12
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
[Kasterisk1*CLI>
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
--- (14 headers 18 lines) ---
Destroying call '79f1c5c80cfc56d5759ed08b150075a0@208.75.208.90'
[Kasterisk1*CLI>
12 headers, 0 lines
Reliably Transmitting (NAT) to 64.192.196.12:5060:
OPTIONS sip:64.192.196.12 SIP/2.0
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK6c377d3d;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as095003e4
To: <sip:64.192.196.12>
Contact: <sip:Unknown@208.75.208.90>
Call-ID: 08b9c8cd247560ed739c613f4230a6d2@208.75.208.90
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 09 Nov 2007 01:54:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
[Kasterisk1*CLI>
<-- SIP read from 64.192.196.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK6c377d3d;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as095003e4
To: <sip:64.192.196.12>;tag=29FECAC6-458
Date: Sat, 09 Mar 2002 03:42:43 GMT
Call-ID: 08b9c8cd247560ed739c613f4230a6d2@208.75.208.90
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,resource-priority,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 451
v=0
o=CiscoSystemsSIP-GW-UserAgent 2176 7254 IN IP4 64.192.196.12
s=SIP Call
c=IN IP4 64.192.196.12
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 15 3
c=IN IP4 64.192.196.12
m=image 0 udptl t38
c=IN IP4 64.192.196.12
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
[Kasterisk1*CLI>
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
--- (14 headers 18 lines) ---
[Kasterisk1*CLI>
Destroying call '08b9c8cd247560ed739c613f4230a6d2@208.75.208.90'
[Kasterisk1*CLI>
12 headers, 0 lines
Reliably Transmitting (NAT) to 64.192.196.12:5060:
OPTIONS sip:64.192.196.12 SIP/2.0
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK22990c5f;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as2edd3f79
To: <sip:64.192.196.12>
Contact: <sip:Unknown@208.75.208.90>
Call-ID: 3096d0f229a9ca285c5e40251fab4007@208.75.208.90
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 09 Nov 2007 01:54:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
[Kasterisk1*CLI>
<-- SIP read from 64.192.196.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK22990c5f;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as2edd3f79
To: <sip:64.192.196.12>;tag=29FECAFE-26A8
Date: Sat, 09 Mar 2002 03:42:43 GMT
Call-ID: 3096d0f229a9ca285c5e40251fab4007@208.75.208.90
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,resource-priority,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 451
v=0
o=CiscoSystemsSIP-GW-UserAgent 1012 3275 IN IP4 64.192.196.12
s=SIP Call
c=IN IP4 64.192.196.12
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 15 3
c=IN IP4 64.192.196.12
m=image 0 udptl t38
c=IN IP4 64.192.196.12
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
--- (14 headers 18 lines) ---
[Kasterisk1*CLI>
Destroying call '3096d0f229a9ca285c5e40251fab4007@208.75.208.90'
[Kasterisk1*CLI>
12 headers, 0 lines
[Kasterisk1*CLI>
Reliably Transmitting (NAT) to 64.192.196.12:5060:
OPTIONS sip:64.192.196.12 SIP/2.0
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK176c2caf;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as1511fa5c
To: <sip:64.192.196.12>
Contact: <sip:Unknown@208.75.208.90>
Call-ID: 6a1c520d3a5bf05a41b47c3b3aa3defc@208.75.208.90
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 09 Nov 2007 01:55:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
[Kasterisk1*CLI>
<-- SIP read from 64.192.196.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK176c2caf;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as1511fa5c
To: <sip:64.192.196.12>;tag=29FFB555-1EF5
Date: Sat, 09 Mar 2002 03:43:43 GMT
Call-ID: 6a1c520d3a5bf05a41b47c3b3aa3defc@208.75.208.90
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,resource-priority,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 451
v=0
o=CiscoSystemsSIP-GW-UserAgent 9717 7781 IN IP4 64.192.196.12
s=SIP Call
c=IN IP4 64.192.196.12
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 15 3
c=IN IP4 64.192.196.12
m=image 0 udptl t38
c=IN IP4 64.192.196.12
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
--- (14 headers 18 lines) ---
[Kasterisk1*CLI>
Destroying call '6a1c520d3a5bf05a41b47c3b3aa3defc@208.75.208.90'
[Kasterisk1*CLI>
12 headers, 0 lines
[Kasterisk1*CLI>
Reliably Transmitting (NAT) to 64.192.196.12:5060:
OPTIONS sip:64.192.196.12 SIP/2.0
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK31983372;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as35318511
To: <sip:64.192.196.12>
Contact: <sip:Unknown@208.75.208.90>
Call-ID: 0faa04a915f1fe9f3b470cd14dd3bebe@208.75.208.90
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 09 Nov 2007 01:55:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
[Kasterisk1*CLI>
<-- SIP read from 64.192.196.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK31983372;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as35318511
To: <sip:64.192.196.12>;tag=29FFB591-922
Date: Sat, 09 Mar 2002 03:43:43 GMT
Call-ID: 0faa04a915f1fe9f3b470cd14dd3bebe@208.75.208.90
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,resource-priority,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 451
v=0
o=CiscoSystemsSIP-GW-UserAgent 4299 3279 IN IP4 64.192.196.12
s=SIP Call
c=IN IP4 64.192.196.12
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 15 3
c=IN IP4 64.192.196.12
m=image 0 udptl t38
c=IN IP4 64.192.196.12
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
[Kasterisk1*CLI>
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
--- (14 headers 18 lines) ---
[Kasterisk1*CLI>
Destroying call '0faa04a915f1fe9f3b470cd14dd3bebe@208.75.208.90'
[Kasterisk1*CLI>
12 headers, 0 lines
Reliably Transmitting (NAT) to 64.192.196.12:5060:
OPTIONS sip:64.192.196.12 SIP/2.0
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK46628080;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as2b703ec5
To: <sip:64.192.196.12>
Contact: <sip:Unknown@208.75.208.90>
Call-ID: 3814e2230ca80cbe71ac54dd33ab758c@208.75.208.90
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 09 Nov 2007 01:56:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
[Kasterisk1*CLI>
<-- SIP read from 64.192.196.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK46628080;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as2b703ec5
To: <sip:64.192.196.12>;tag=2A009FE7-2B5
Date: Sat, 09 Mar 2002 03:44:43 GMT
Call-ID: 3814e2230ca80cbe71ac54dd33ab758c@208.75.208.90
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,resource-priority,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 450
v=0
o=CiscoSystemsSIP-GW-UserAgent 584 8940 IN IP4 64.192.196.12
s=SIP Call
c=IN IP4 64.192.196.12
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 15 3
c=IN IP4 64.192.196.12
m=image 0 udptl t38
c=IN IP4 64.192.196.12
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a
[Kasterisk1*CLI>
=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
--- (14 headers 18 lines) ---
[Kasterisk1*CLI>
Destroying call '3814e2230ca80cbe71ac54dd33ab758c@208.75.208.90'
[Kasterisk1*CLI>
12 headers, 0 lines
[Kasterisk1*CLI>
Reliably Transmitting (NAT) to 64.192.196.12:5060:
OPTIONS sip:64.192.196.12 SIP/2.0
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK572570a3;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as78a44f10
To: <sip:64.192.196.12>
Contact: <sip:Unknown@208.75.208.90>
Call-ID: 3221e29f298db43e6644e49d713784a7@208.75.208.90
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 09 Nov 2007 01:56:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
[Kasterisk1*CLI>
<-- SIP read from 64.192.196.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK572570a3;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as78a44f10
To: <sip:64.192.196.12>;tag=2A00A01F-DEE
Date: Sat, 09 Mar 2002 03:44:44 GMT
Call-ID: 3221e29f298db43e6644e49d713784a7@208.75.208.90
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,resource-priority,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 451
v=0
o=CiscoSystemsSIP-GW-UserAgent 8052 7088 IN IP4 64.192.196.12
s=SIP Call
c=IN IP4 64.192.196.12
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 15 3
c=IN IP4 64.192.196.12
m=image 0 udptl t38
c=IN IP4 64.192.196.12
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
[Kasterisk1*CLI>
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
--- (14 headers 18 lines) ---
[Kasterisk1*CLI>
Destroying call '3221e29f298db43e6644e49d713784a7@208.75.208.90'
[Kasterisk1*CLI>
12 headers, 0 lines
Reliably Transmitting (NAT) to 64.192.196.12:5060:
OPTIONS sip:64.192.196.12 SIP/2.0
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK18709ddf;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as0136b16e
To: <sip:64.192.196.12>
Contact: <sip:Unknown@208.75.208.90>
Call-ID: 1443c495660d77202eb388c64a72a0a4@208.75.208.90
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 09 Nov 2007 01:57:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
[Kasterisk1*CLI>
<-- SIP read from 64.192.196.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK18709ddf;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as0136b16e
To: <sip:64.192.196.12>;tag=2A018A76-618
Date: Sat, 09 Mar 2002 03:45:44 GMT
Call-ID: 1443c495660d77202eb388c64a72a0a4@208.75.208.90
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,resource-priority,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 451
v=0
o=CiscoSystemsSIP-GW-UserAgent 8714 7159 IN IP4 64.192.196.12
s=SIP Call
c=IN IP4 64.192.196.12
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 15 3
c=IN IP4 64.192.196.12
m=image 0 udptl t38
c=IN IP4 64.192.196.12
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
[Kasterisk1*CLI>
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
--- (14 headers 18 lines) ---
[Kasterisk1*CLI>
Destroying call '1443c495660d77202eb388c64a72a0a4@208.75.208.90'
[Kasterisk1*CLI>
12 headers, 0 lines
Reliably Transmitting (NAT) to 64.192.196.12:5060:
OPTIONS sip:64.192.196.12 SIP/2.0
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK4bd9690f;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as247e3211
To: <sip:64.192.196.12>
Contact: <sip:Unknown@208.75.208.90>
Call-ID: 270054c4530a72fa528874c063718a59@208.75.208.90
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 09 Nov 2007 01:57:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
[Kasterisk1*CLI>
<-- SIP read from 64.192.196.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK4bd9690f;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as247e3211
To: <sip:64.192.196.12>;tag=2A018AAE-196B
Date: Sat, 09 Mar 2002 03:45:44 GMT
Call-ID: 270054c4530a72fa528874c063718a59@208.75.208.90
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,resource-priority,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 451
v=0
o=CiscoSystemsSIP-GW-UserAgent 8871 5410 IN IP4 64.192.196.12
s=SIP Call
c=IN IP4 64.192.196.12
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 15 3
c=IN IP4 64.192.196.12
m=image 0 udptl t38
c=IN IP4 64.192.196.12
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
--- (14 headers 18 lines) ---
[Kasterisk1*CLI>
Destroying call '270054c4530a72fa528874c063718a59@208.75.208.90'
[Kasterisk1*CLI>
12 headers, 0 lines
Reliably Transmitting (NAT) to 64.192.196.12:5060:
OPTIONS sip:64.192.196.12 SIP/2.0
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK0b1e3ac9;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as004ade23
To: <sip:64.192.196.12>
Contact: <sip:Unknown@208.75.208.90>
Call-ID: 70a45ac142de29e06add3bdb573864be@208.75.208.90
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 09 Nov 2007 01:58:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
[Kasterisk1*CLI>
<-- SIP read from 64.192.196.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK0b1e3ac9;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as004ade23
To: <sip:64.192.196.12>;tag=2A027504-4FB
Date: Sat, 09 Mar 2002 03:46:44 GMT
Call-ID: 70a45ac142de29e06add3bdb573864be@208.75.208.90
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,resource-priority,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 451
v=0
o=CiscoSystemsSIP-GW-UserAgent 2915 9848 IN IP4 64.192.196.12
s=SIP Call
c=IN IP4 64.192.196.12
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 15 3
c=IN IP4 64.192.196.12
m=image 0 udptl t38
c=IN IP4 64.192.196.12
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
[Kasterisk1*CLI>
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
--- (14 headers 18 lines) ---
[Kasterisk1*CLI>
Destroying call '70a45ac142de29e06add3bdb573864be@208.75.208.90'
[Kasterisk1*CLI>
12 headers, 0 lines
Reliably Transmitting (NAT) to 64.192.196.12:5060:
OPTIONS sip:64.192.196.12 SIP/2.0
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK3b8ae08d;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as3fd9c9b1
To: <sip:64.192.196.12>
Contact: <sip:Unknown@208.75.208.90>
Call-ID: 2b5685df63632ec565d2f0121c435e18@208.75.208.90
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 09 Nov 2007 01:58:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
[Kasterisk1*CLI>
<-- SIP read from 64.192.196.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK3b8ae08d;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as3fd9c9b1
To: <sip:64.192.196.12>;tag=2A02753C-6DA
Date: Sat, 09 Mar 2002 03:46:44 GMT
Call-ID: 2b5685df63632ec565d2f0121c435e18@208.75.208.90
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,resource-priority,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 451
v=0
o=CiscoSystemsSIP-GW-UserAgent 7820 8621 IN IP4 64.192.196.12
s=SIP Call
c=IN IP4 64.192.196.12
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 15 3
c=IN IP4 64.192.196.12
m=image 0 udptl t38
c=IN IP4 64.192.196.12
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
[Kasterisk1*CLI>
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
--- (14 headers 18 lines) ---
[Kasterisk1*CLI>
Destroying call '2b5685df63632ec565d2f0121c435e18@208.75.208.90'
[Kasterisk1*CLI>
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/5060-b7219088", "OUTNUM=9082296706") in new stack
-- Executing Set("SIP/5060-b7219088", "custom=IAX2/GTT_No_VPN") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "0?customtrunk") in new stack
-- Executing Set("SIP/5060-b7219088", "CDR(userfield)=204.14.9.50") in new stack
-- Executing Dial("SIP/5060-b7219088", "IAX2/GTT_No_VPN/9082296706|300|") in new stack
-- Called GTT_No_VPN/9082296706
[Kasterisk1*CLI>
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/5060-b72dd5d8", "OUTNUM=9086840826") in new stack
-- Executing Set("SIP/5060-b72dd5d8", "custom=IAX2/VPN") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/5060-b72dd5d8", "0?customtrunk") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b72dd5d8", "CDR(userfield)=204.14.9.50") in new stack
[Kasterisk1*CLI>
-- Executing Dial("SIP/5060-b72dd5d8", "IAX2/VPN/9086840826|300|") in new stack
[Kasterisk1*CLI>
== Everyone is busy/congested at this time (1:0/0/1)
[Kasterisk1*CLI>
-- Executing Goto("SIP/5060-b72dd5d8", "s-CHANUNAVAIL|1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing NoOp("SIP/5060-b72dd5d8", "Dial failed due to CHANUNAVAIL - failing through to other trunks") in new stack
-- Executing Macro("SIP/5060-b72dd5d8", "outisbusy|") in new stack
-- Executing Playback("SIP/5060-b72dd5d8", "all-circuits-busy-now|noanswer") in new stack
-- Playing 'all-circuits-busy-now' (language 'en')
[Kasterisk1*CLI>
-- Call accepted by 190.80.53.168 (format g723)
-- Format for call is g723
[Kasterisk1*CLI>
-- Executing Playback("SIP/5060-b72dd5d8", "pls-try-call-later|noanswer") in new stack
[Kasterisk1*CLI>
-- Playing 'pls-try-call-later' (language 'en')
[Kasterisk1*CLI>
-- IAX2/GTT_No_VPN-16392 is ringing
[Kasterisk1*CLI>
-- Hungup 'IAX2/GTT_No_VPN-16386'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/5060-b7229810' in macro 'dialout-trunk'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/5060-b7229810'
-- Executing Macro("SIP/5060-b7229810", "hangupcall") in new stack
-- Executing ResetCDR("SIP/5060-b7229810", "w") in new stack
[Kasterisk1*CLI>
-- Executing NoCDR("SIP/5060-b7229810", "") in new stack
-- Executing GotoIf("SIP/5060-b7229810", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing GotoIf("SIP/5060-b7229810", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing Wait("SIP/5060-b7229810", "5") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/5060-b7229810' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/5060-b7229810'
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b72dd5d8", "CDR(userfield)=204.14.9.50") in new stack
-- Executing Macro("SIP/5060-b72dd5d8", "hangupcall") in new stack
-- Executing ResetCDR("SIP/5060-b72dd5d8", "w") in new stack
[Kasterisk1*CLI>
-- Executing NoCDR("SIP/5060-b72dd5d8", "") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/5060-b72dd5d8", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/5060-b72dd5d8", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing Wait("SIP/5060-b72dd5d8", "5") in new stack
[Kasterisk1*CLI>
-- Executing Macro("SIP/204.14.9.50-b720cfe0", "dialout-trunk|10|9082207434||") in new stack
-- Executing Set("SIP/204.14.9.50-b720cfe0", "CDR(userfield)=204.14.9.50") in new stack
-- Executing Set("SIP/204.14.9.50-b720cfe0", "DIAL_TRUNK=10") in new stack
-- Executing Set("SIP/204.14.9.50-b720cfe0", "_NODEST=") in new stack
-- Executing Set("SIP/204.14.9.50-b720cfe0", "DIAL_NUMBER=9082207434") in new stack
-- Executing Set("SIP/204.14.9.50-b720cfe0", "ROUTE_PASSWD=") in new stack
-- Executing Set("SIP/204.14.9.50-b720cfe0", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing GotoIf("SIP/204.14.9.50-b720cfe0", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing Set("SIP/204.14.9.50-b720cfe0", "GROUP()=OUT_10") in new stack
-- Executing Macro("SIP/204.14.9.50-b720cfe0", "user-callerid|SKIPTTL") in new stack
-- Executing NoOp("SIP/204.14.9.50-b720cfe0", "user-callerid: 9006") in new stack
-- Executing GotoIf("SIP/204.14.9.50-b720cfe0", "0?report") in new stack
-- Executing GotoIf("SIP/204.14.9.50-b720cfe0", "0?start") in new stack
-- Executing Set("SIP/204.14.9.50-b720cfe0", "REALCALLERIDNUM=9006") in new stack
-- Executing NoOp("SIP/204.14.9.50-b720cfe0", "REALCALLERIDNUM is 9006") in new stack
-- Executing Set("SIP/204.14.9.50-b720cfe0", "AMPUSER=") in new stack
-- Executing Set("SIP/204.14.9.50-b720cfe0", "AMPUSERCIDNAME=") in new stack
-- Executing GotoIf("SIP/204.14.9.50-b720cfe0", "1?report") in new stack
-- Goto (macro-user-callerid,s,11)
-- Executing NoOp("SIP/204.14.9.50-b720cfe0", "TTL: ARG1: SKIPTTL") in new stack
-- Executing GotoIf("SIP/204.14.9.50-b720cfe0", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/204.14.9.50-b720cfe0", "Using CallerID "" <9006>") in new stack
-- Executing Macro("SIP/204.14.9.50-b720cfe0", "record-enable|9006|OUT") in new stack
-- Executing GotoIf("SIP/204.14.9.50-b720cfe0", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("SIP/204.14.9.50-b720cfe0", "recordingcheck|20071109-015836|1194573516.135398") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
[Kasterisk1*CLI>
recordingcheck|20071109-015836|1194573516.135398: No AMPUSER db entry for 9006. Not recording
[Kasterisk1*CLI>
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/204.14.9.50-b720cfe0", "No recording needed") in new stack
-- Executing GotoIf("SIP/204.14.9.50-b720cfe0", "0?skipoutcid") in new stack
-- Executing Set("SIP/204.14.9.50-b720cfe0", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing Macro("SIP/204.14.9.50-b720cfe0", "outbound-callerid|10") in new stack
-- Executing GotoIf("SIP/204.14.9.50-b720cfe0", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing NoOp("SIP/204.14.9.50-b720cfe0", "REALCALLERIDNUM is 9006") in new stack
-- Executing GotoIf("SIP/204.14.9.50-b720cfe0", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing Set("SIP/204.14.9.50-b720cfe0", "USEROUTCID=") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/204.14.9.50-b720cfe0", "EMERGENCYCID=") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/204.14.9.50-b720cfe0", "TRUNKOUTCID=") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/204.14.9.50-b720cfe0", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/204.14.9.50-b720cfe0", "1?usercid") in new stack
[Kasterisk1*CLI>
-- Goto (macro-outbound-callerid,s,18)
-- Executing GotoIf("SIP/204.14.9.50-b720cfe0", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing NoOp("SIP/204.14.9.50-b720cfe0", "CallerID set to "" <9006>") in new stack
-- Executing GotoIf("SIP/204.14.9.50-b720cfe0", "0?nomax") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/204.14.9.50-b720cfe0", "0?chanfull") in new stack
[Kasterisk1*CLI>
-- Executing DeadAGI("SIP/204.14.9.50-b720cfe0", "fixlocalprefix") in new stack
[Kasterisk1*CLI>
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
[Kasterisk1*CLI>
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/204.14.9.50-b720cfe0", "OUTNUM=9082207434") in new stack
-- Executing Set("SIP/204.14.9.50-b720cfe0", "custom=IAX2/GTT_No_VPN") in new stack
-- Executing GotoIf("SIP/204.14.9.50-b720cfe0", "0?customtrunk") in new stack
-- Executing Set("SIP/204.14.9.50-b720cfe0", "CDR(userfield)=204.14.9.50") in new stack
-- Executing Dial("SIP/204.14.9.50-b720cfe0", "IAX2/GTT_No_VPN/9082207434|300|") in new stack
-- Called GTT_No_VPN/9082207434
[Kasterisk1*CLI>
-- Call accepted by 190.80.53.168 (format g723)
-- Format for call is g723
[Kasterisk1*CLI>
-- Hungup 'IAX2/GTT_No_VPN-16417'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/66.246.197.20-b70257d8' in macro 'dialout-trunk'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/66.246.197.20-b70257d8'
-- Executing Macro("SIP/66.246.197.20-b70257d8", "hangupcall") in new stack
-- Executing ResetCDR("SIP/66.246.197.20-b70257d8", "w") in new stack
[Kasterisk1*CLI>
-- Executing NoCDR("SIP/66.246.197.20-b70257d8", "") in new stack
-- Executing GotoIf("SIP/66.246.197.20-b70257d8", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing GotoIf("SIP/66.246.197.20-b70257d8", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing Wait("SIP/66.246.197.20-b70257d8", "5") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/66.246.197.20-b70257d8' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/66.246.197.20-b70257d8'
[Kasterisk1*CLI>
-- Hungup 'IAX2/GTT_No_VPN-16388'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/66.246.197.20-b729f710' in macro 'dialout-trunk'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/66.246.197.20-b729f710'
-- Executing Macro("SIP/66.246.197.20-b729f710", "hangupcall") in new stack
-- Executing ResetCDR("SIP/66.246.197.20-b729f710", "w") in new stack
[Kasterisk1*CLI>
-- Executing NoCDR("SIP/66.246.197.20-b729f710", "") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/66.246.197.20-b729f710", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing GotoIf("SIP/66.246.197.20-b729f710", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing Wait("SIP/66.246.197.20-b729f710", "5") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/66.246.197.20-b729f710' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/66.246.197.20-b729f710'
== End MixMonitor Recording SIP/66.246.197.20-b729f710
[Kasterisk1*CLI>
-- Executing Macro("SIP/64.15.135.252-b729f6f8", "dialout-trunk|17|9086850715||") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/64.15.135.252-b729f6f8", "CDR(userfield)=64.15.135.252") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/64.15.135.252-b729f6f8", "DIAL_TRUNK=17") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/64.15.135.252-b729f6f8", "_NODEST=") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/64.15.135.252-b729f6f8", "DIAL_NUMBER=9086850715") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/64.15.135.252-b729f6f8", "ROUTE_PASSWD=") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/64.15.135.252-b729f6f8", "DIAL_TRUNK_OPTIONS=tr") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/64.15.135.252-b729f6f8", "1?noauth") in new stack
[Kasterisk1*CLI>
-- Goto (macro-dialout-trunk,s,9)
[Kasterisk1*CLI>
-- Executing Set("SIP/64.15.135.252-b729f6f8", "GROUP()=OUT_17") in new stack
[Kasterisk1*CLI>
-- Executing Macro("SIP/64.15.135.252-b729f6f8", "user-callerid|SKIPTTL") in new stack
[Kasterisk1*CLI>
-- Executing NoOp("SIP/64.15.135.252-b729f6f8", "user-callerid: 7aaa 9015") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b729f6f8", "0?report") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b729f6f8", "0?start") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/64.15.135.252-b729f6f8", "REALCALLERIDNUM=9015") in new stack
-- Executing NoOp("SIP/64.15.135.252-b729f6f8", "REALCALLERIDNUM is 9015") in new stack
-- Executing Set("SIP/64.15.135.252-b729f6f8", "AMPUSER=") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/64.15.135.252-b729f6f8", "AMPUSERCIDNAME=") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/64.15.135.252-b729f6f8", "1?report") in new stack
[Kasterisk1*CLI>
-- Goto (macro-user-callerid,s,11)
[Kasterisk1*CLI>
-- Executing NoOp("SIP/64.15.135.252-b729f6f8", "TTL: ARG1: SKIPTTL") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/64.15.135.252-b729f6f8", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/64.15.135.252-b729f6f8", "Using CallerID "7aaa" <9015>") in new stack
[Kasterisk1*CLI>
-- Executing Macro("SIP/64.15.135.252-b729f6f8", "record-enable|9015|OUT") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b729f6f8", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
[Kasterisk1*CLI>
-- Executing DeadAGI("SIP/64.15.135.252-b729f6f8", "recordingcheck|20071109-015838|1194573518.135400") in new stack
[Kasterisk1*CLI>
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
[Kasterisk1*CLI>
recordingcheck|20071109-015838|1194573518.135400: No AMPUSER db entry for 9015. Not recording
[Kasterisk1*CLI>
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/64.15.135.252-b729f6f8", "No recording needed") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b729f6f8", "0?skipoutcid") in new stack
-- Executing Set("SIP/64.15.135.252-b729f6f8", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing Macro("SIP/64.15.135.252-b729f6f8", "outbound-callerid|17") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b729f6f8", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing NoOp("SIP/64.15.135.252-b729f6f8", "REALCALLERIDNUM is 9015") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b729f6f8", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing Set("SIP/64.15.135.252-b729f6f8", "USEROUTCID=") in new stack
-- Executing Set("SIP/64.15.135.252-b729f6f8", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/64.15.135.252-b729f6f8", "TRUNKOUTCID=") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b729f6f8", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing GotoIf("SIP/64.15.135.252-b729f6f8", "1?usercid") in new stack
-- Goto (macro-outbound-callerid,s,18)
-- Executing GotoIf("SIP/64.15.135.252-b729f6f8", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing NoOp("SIP/64.15.135.252-b729f6f8", "CallerID set to "7aaa" <9015>") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b729f6f8", "0?nomax") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b729f6f8", "0?chanfull") in new stack
-- Executing DeadAGI("SIP/64.15.135.252-b729f6f8", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
[Kasterisk1*CLI>
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/64.15.135.252-b729f6f8", "OUTNUM=9086850715") in new stack
-- Executing Set("SIP/64.15.135.252-b729f6f8", "custom=IAX2/Vydall") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b729f6f8", "0?customtrunk") in new stack
-- Executing Set("SIP/64.15.135.252-b729f6f8", "CDR(userfield)=64.15.135.252") in new stack
-- Executing Dial("SIP/64.15.135.252-b729f6f8", "IAX2/Vydall/9086850715|300|") in new stack
-- Called Vydall/9086850715
[Kasterisk1*CLI>
-- Call accepted by 190.80.53.205 (format g723)
-- Format for call is g723
[Kasterisk1*CLI>
-- Hungup 'IAX2/GTT_No_VPN-16384'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/5060-b722ed50' in macro 'dialout-trunk'
[Kasterisk1*CLI>
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/5060-b722ed50'
[Kasterisk1*CLI>
-- Executing Macro("SIP/5060-b722ed50", "hangupcall") in new stack
[Kasterisk1*CLI>
-- Executing ResetCDR("SIP/5060-b722ed50", "w") in new stack
[Kasterisk1*CLI>
-- Executing NoCDR("SIP/5060-b722ed50", "") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/5060-b722ed50", "1?skiprg") in new stack
[Kasterisk1*CLI>
-- Goto (macro-hangupcall,s,6)
-- Executing GotoIf("SIP/5060-b722ed50", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing Wait("SIP/5060-b722ed50", "5") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/5060-b722ed50' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/5060-b722ed50'
[Kasterisk1*CLI>
-- IAX2/Vydall-16399 is ringing
[Kasterisk1*CLI>
-- Executing Macro("SIP/5060-b729c010", "dialout-trunk|10|9082277487||") in new stack
-- Executing Set("SIP/5060-b729c010", "CDR(userfield)=70.42.39.67") in new stack
-- Executing Set("SIP/5060-b729c010", "DIAL_TRUNK=10") in new stack
-- Executing Set("SIP/5060-b729c010", "_NODEST=") in new stack
-- Executing Set("SIP/5060-b729c010", "DIAL_NUMBER=9082277487") in new stack
-- Executing Set("SIP/5060-b729c010", "ROUTE_PASSWD=") in new stack
-- Executing Set("SIP/5060-b729c010", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing GotoIf("SIP/5060-b729c010", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing Set("SIP/5060-b729c010", "GROUP()=OUT_10") in new stack
-- Executing Macro("SIP/5060-b729c010", "user-callerid|SKIPTTL") in new stack
-- Executing NoOp("SIP/5060-b729c010", "user-callerid: Rapidlink 9010") in new stack
-- Executing GotoIf("SIP/5060-b729c010", "0?report") in new stack
-- Executing GotoIf("SIP/5060-b729c010", "0?start") in new stack
-- Executing Set("SIP/5060-b729c010", "REALCALLERIDNUM=9010") in new stack
-- Executing NoOp("SIP/5060-b729c010", "REALCALLERIDNUM is 9010") in new stack
-- Executing Set("SIP/5060-b729c010", "AMPUSER=") in new stack
-- Executing Set("SIP/5060-b729c010", "AMPUSERCIDNAME=") in new stack
-- Executing GotoIf("SIP/5060-b729c010", "1?report") in new stack
-- Goto (macro-user-callerid,s,11)
-- Executing NoOp("SIP/5060-b729c010", "TTL: ARG1: SKIPTTL") in new stack
-- Executing GotoIf("SIP/5060-b729c010", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/5060-b729c010", "Using CallerID "Rapidlink" <9010>") in new stack
-- Executing Macro("SIP/5060-b729c010", "record-enable|9010|OUT") in new stack
-- Executing GotoIf("SIP/5060-b729c010", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("SIP/5060-b729c010", "recordingcheck|20071109-015840|1194573520.135402") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
[Kasterisk1*CLI>
recordingcheck|20071109-015840|1194573520.135402: No AMPUSER db entry for 9010. Not recording
[Kasterisk1*CLI>
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/5060-b729c010", "No recording needed") in new stack
-- Executing GotoIf("SIP/5060-b729c010", "0?skipoutcid") in new stack
-- Executing Set("SIP/5060-b729c010", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing Macro("SIP/5060-b729c010", "outbound-callerid|10") in new stack
-- Executing GotoIf("SIP/5060-b729c010", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing NoOp("SIP/5060-b729c010", "REALCALLERIDNUM is 9010") in new stack
-- Executing GotoIf("SIP/5060-b729c010", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing Set("SIP/5060-b729c010", "USEROUTCID=") in new stack
-- Executing Set("SIP/5060-b729c010", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/5060-b729c010", "TRUNKOUTCID=") in new stack
-- Executing GotoIf("SIP/5060-b729c010", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing GotoIf("SIP/5060-b729c010", "1?usercid") in new stack
-- Goto (macro-outbound-callerid,s,18)
-- Executing GotoIf("SIP/5060-b729c010", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing NoOp("SIP/5060-b729c010", "CallerID set to "Rapidlink" <9010>") in new stack
-- Executing GotoIf("SIP/5060-b729c010", "0?nomax") in new stack
-- Executing GotoIf("SIP/5060-b729c010", "0?chanfull") in new stack
-- Executing DeadAGI("SIP/5060-b729c010", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
[Kasterisk1*CLI>
-- AGI Script fixlocalprefix completed, returning 0
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b729c010", "OUTNUM=9082277487") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b729c010", "custom=IAX2/GTT_No_VPN") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/5060-b729c010", "0?customtrunk") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b729c010", "CDR(userfield)=70.42.39.67") in new stack
[Kasterisk1*CLI>
-- Executing Dial("SIP/5060-b729c010", "IAX2/GTT_No_VPN/9082277487|300|") in new stack
[Kasterisk1*CLI>
-- Called GTT_No_VPN/9082277487
[Kasterisk1*CLI>
-- Executing Macro("SIP/5060-b7243938", "dialout-trunk|10|9082683020||") in new stack
-- Executing Set("SIP/5060-b7243938", "CDR(userfield)=204.14.9.50") in new stack
-- Executing Set("SIP/5060-b7243938", "DIAL_TRUNK=10") in new stack
-- Executing Set("SIP/5060-b7243938", "_NODEST=") in new stack
-- Executing Set("SIP/5060-b7243938", "DIAL_NUMBER=9082683020") in new stack
-- Executing Set("SIP/5060-b7243938", "ROUTE_PASSWD=") in new stack
-- Executing Set("SIP/5060-b7243938", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing GotoIf("SIP/5060-b7243938", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing Set("SIP/5060-b7243938", "GROUP()=OUT_10") in new stack
-- Executing Macro("SIP/5060-b7243938", "user-callerid|SKIPTTL") in new stack
-- Executing NoOp("SIP/5060-b7243938", "user-callerid: ") in new stack
-- Executing GotoIf("SIP/5060-b7243938", "0?report") in new stack
-- Executing GotoIf("SIP/5060-b7243938", "0?start") in new stack
-- Executing Set("SIP/5060-b7243938", "REALCALLERIDNUM=") in new stack
-- Executing NoOp("SIP/5060-b7243938", "REALCALLERIDNUM is ") in new stack
-- Executing Set("SIP/5060-b7243938", "AMPUSER=") in new stack
-- Executing Set("SIP/5060-b7243938", "AMPUSERCIDNAME=") in new stack
-- Executing GotoIf("SIP/5060-b7243938", "1?report") in new stack
-- Goto (macro-user-callerid,s,11)
-- Executing NoOp("SIP/5060-b7243938", "TTL: ARG1: SKIPTTL") in new stack
-- Executing GotoIf("SIP/5060-b7243938", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/5060-b7243938", "Using CallerID "" <>") in new stack
-- Executing Macro("SIP/5060-b7243938", "record-enable||OUT") in new stack
-- Executing GotoIf("SIP/5060-b7243938", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("SIP/5060-b7243938", "recordingcheck|20071109-015840|1194573520.135404") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
[Kasterisk1*CLI>
-- Call accepted by 190.80.53.168 (format g723)
-- Format for call is g723
[Kasterisk1*CLI>
recordingcheck|20071109-015840|1194573520.135404: No AMPUSER db entry for . Not recording
[Kasterisk1*CLI>
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/5060-b7243938", "No recording needed") in new stack
-- Executing GotoIf("SIP/5060-b7243938", "0?skipoutcid") in new stack
-- Executing Set("SIP/5060-b7243938", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing Macro("SIP/5060-b7243938", "outbound-callerid|10") in new stack
-- Executing GotoIf("SIP/5060-b7243938", "0?start") in new stack
-- Executing Set("SIP/5060-b7243938", "REALCALLERIDNUM=") in new stack
-- Executing NoOp("SIP/5060-b7243938", "REALCALLERIDNUM is ") in new stack
-- Executing GotoIf("SIP/5060-b7243938", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing Set("SIP/5060-b7243938", "USEROUTCID=") in new stack
-- Executing Set("SIP/5060-b7243938", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/5060-b7243938", "TRUNKOUTCID=") in new stack
-- Executing GotoIf("SIP/5060-b7243938", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing GotoIf("SIP/5060-b7243938", "1?usercid") in new stack
-- Goto (macro-outbound-callerid,s,18)
-- Executing GotoIf("SIP/5060-b7243938", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing NoOp("SIP/5060-b7243938", "CallerID set to "" <>") in new stack
-- Executing GotoIf("SIP/5060-b7243938", "0?nomax") in new stack
-- Executing GotoIf("SIP/5060-b7243938", "0?chanfull") in new stack
-- Executing DeadAGI("SIP/5060-b7243938", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
[Kasterisk1*CLI>
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/5060-b7243938", "OUTNUM=9082683020") in new stack
-- Executing Set("SIP/5060-b7243938", "custom=IAX2/GTT_No_VPN") in new stack
-- Executing GotoIf("SIP/5060-b7243938", "0?customtrunk") in new stack
-- Executing Set("SIP/5060-b7243938", "CDR(userfield)=204.14.9.50") in new stack
-- Executing Dial("SIP/5060-b7243938", "IAX2/GTT_No_VPN/9082683020|300|") in new stack
-- Called GTT_No_VPN/9082683020
[Kasterisk1*CLI>
-- Call accepted by 190.80.53.168 (format g723)
-- Format for call is g723
[Kasterisk1*CLI>
-- Executing Macro("SIP/64.15.135.252-b706d540", "dialout-trunk|10|9086260284||") in new stack
-- Executing Set("SIP/64.15.135.252-b706d540", "CDR(userfield)=64.15.135.252") in new stack
-- Executing Set("SIP/64.15.135.252-b706d540", "DIAL_TRUNK=10") in new stack
-- Executing Set("SIP/64.15.135.252-b706d540", "_NODEST=") in new stack
-- Executing Set("SIP/64.15.135.252-b706d540", "DIAL_NUMBER=9086260284") in new stack
-- Executing Set("SIP/64.15.135.252-b706d540", "ROUTE_PASSWD=") in new stack
-- Executing Set("SIP/64.15.135.252-b706d540", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b706d540", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing Set("SIP/64.15.135.252-b706d540", "GROUP()=OUT_10") in new stack
-- Executing Macro("SIP/64.15.135.252-b706d540", "user-callerid|SKIPTTL") in new stack
-- Executing NoOp("SIP/64.15.135.252-b706d540", "user-callerid: 7aaa 9015") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b706d540", "0?report") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b706d540", "0?start") in new stack
-- Executing Set("SIP/64.15.135.252-b706d540", "REALCALLERIDNUM=9015") in new stack
-- Executing NoOp("SIP/64.15.135.252-b706d540", "REALCALLERIDNUM is 9015") in new stack
-- Executing Set("SIP/64.15.135.252-b706d540", "AMPUSER=") in new stack
-- Executing Set("SIP/64.15.135.252-b706d540", "AMPUSERCIDNAME=") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b706d540", "1?report") in new stack
-- Goto (macro-user-callerid,s,11)
-- Executing NoOp("SIP/64.15.135.252-b706d540", "TTL: ARG1: SKIPTTL") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b706d540", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/64.15.135.252-b706d540", "Using CallerID "7aaa" <9015>") in new stack
-- Executing Macro("SIP/64.15.135.252-b706d540", "record-enable|9015|OUT") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b706d540", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("SIP/64.15.135.252-b706d540", "recordingcheck|20071109-015841|1194573521.135406") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
[Kasterisk1*CLI>
recordingcheck|20071109-015841|1194573521.135406: No AMPUSER db entry for 9015. Not recording
[Kasterisk1*CLI>
-- Executing Hangup("SIP/5060-b72dd5d8", "") in new stack
[Kasterisk1*CLI>
== Spawn extension (macro-hangupcall, s, 10) exited non-zero on 'SIP/5060-b72dd5d8' in macro 'hangupcall'
[Kasterisk1*CLI>
== Spawn extension (macro-hangupcall, s, 10) exited non-zero on 'SIP/5060-b72dd5d8' in macro 'outisbusy'
== Spawn extension (macro-hangupcall, s, 10) exited non-zero on 'SIP/5060-b72dd5d8'
[Kasterisk1*CLI>
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/64.15.135.252-b706d540", "No recording needed") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b706d540", "0?skipoutcid") in new stack
-- Executing Set("SIP/64.15.135.252-b706d540", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing Macro("SIP/64.15.135.252-b706d540", "outbound-callerid|10") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b706d540", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing NoOp("SIP/64.15.135.252-b706d540", "REALCALLERIDNUM is 9015") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b706d540", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing Set("SIP/64.15.135.252-b706d540", "USEROUTCID=") in new stack
-- Executing Set("SIP/64.15.135.252-b706d540", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/64.15.135.252-b706d540", "TRUNKOUTCID=") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b706d540", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing GotoIf("SIP/64.15.135.252-b706d540", "1?usercid") in new stack
-- Goto (macro-outbound-callerid,s,18)
-- Executing GotoIf("SIP/64.15.135.252-b706d540", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing NoOp("SIP/64.15.135.252-b706d540", "CallerID set to "7aaa" <9015>") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b706d540", "0?nomax") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b706d540", "0?chanfull") in new stack
-- Executing DeadAGI("SIP/64.15.135.252-b706d540", "fixlocalprefix") in new stack
[Kasterisk1*CLI>
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
[Kasterisk1*CLI>
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/64.15.135.252-b706d540", "OUTNUM=9086260284") in new stack
-- Executing Set("SIP/64.15.135.252-b706d540", "custom=IAX2/GTT_No_VPN") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b706d540", "0?customtrunk") in new stack
-- Executing Set("SIP/64.15.135.252-b706d540", "CDR(userfield)=64.15.135.252") in new stack
-- Executing Dial("SIP/64.15.135.252-b706d540", "IAX2/GTT_No_VPN/9086260284|300|") in new stack
[Kasterisk1*CLI>
-- Called GTT_No_VPN/9086260284
[Kasterisk1*CLI>
-- Hungup 'IAX2/GTT_No_VPN-16420'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/64.15.135.252-b707ae28' in macro 'dialout-trunk'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/64.15.135.252-b707ae28'
-- Executing Macro("SIP/64.15.135.252-b707ae28", "hangupcall") in new stack
-- Executing ResetCDR("SIP/64.15.135.252-b707ae28", "w") in new stack
[Kasterisk1*CLI>
-- Executing NoCDR("SIP/64.15.135.252-b707ae28", "") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b707ae28", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing GotoIf("SIP/64.15.135.252-b707ae28", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing Wait("SIP/64.15.135.252-b707ae28", "5") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/64.15.135.252-b707ae28' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/64.15.135.252-b707ae28'
[Kasterisk1*CLI>
-- Call accepted by 190.80.53.168 (format g723)
-- Format for call is g723
[Kasterisk1*CLI>
-- IAX2/GTT_No_VPN-16387 is ringing
[Kasterisk1*CLI>
-- Hungup 'IAX2/GTT_No_VPN-16404'
[Kasterisk1*CLI>
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/5060-b7219088' in macro 'dialout-trunk'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/5060-b7219088'
-- Executing Macro("SIP/5060-b7219088", "hangupcall") in new stack
-- Executing ResetCDR("SIP/5060-b7219088", "w") in new stack
-- Executing NoCDR("SIP/5060-b7219088", "") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/5060-b7219088", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing Wait("SIP/5060-b7219088", "5") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/5060-b7219088' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/5060-b7219088'
[Kasterisk1*CLI>
-- Executing Macro("SIP/5060-b7219088", "dialout-trunk|10|9082655827||") in new stack
-- Executing Set("SIP/5060-b7219088", "CDR(userfield)=204.14.9.50") in new stack
-- Executing Set("SIP/5060-b7219088", "DIAL_TRUNK=10") in new stack
-- Executing Set("SIP/5060-b7219088", "_NODEST=") in new stack
-- Executing Set("SIP/5060-b7219088", "DIAL_NUMBER=9082655827") in new stack
-- Executing Set("SIP/5060-b7219088", "ROUTE_PASSWD=") in new stack
-- Executing Set("SIP/5060-b7219088", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing Set("SIP/5060-b7219088", "GROUP()=OUT_10") in new stack
-- Executing Macro("SIP/5060-b7219088", "user-callerid|SKIPTTL") in new stack
-- Executing NoOp("SIP/5060-b7219088", "user-callerid: 9006") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "0?report") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "0?start") in new stack
-- Executing Set("SIP/5060-b7219088", "REALCALLERIDNUM=9006") in new stack
-- Executing NoOp("SIP/5060-b7219088", "REALCALLERIDNUM is 9006") in new stack
-- Executing Set("SIP/5060-b7219088", "AMPUSER=") in new stack
-- Executing Set("SIP/5060-b7219088", "AMPUSERCIDNAME=") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "1?report") in new stack
-- Goto (macro-user-callerid,s,11)
-- Executing NoOp("SIP/5060-b7219088", "TTL: ARG1: SKIPTTL") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/5060-b7219088", "Using CallerID "" <9006>") in new stack
-- Executing Macro("SIP/5060-b7219088", "record-enable|9006|OUT") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("SIP/5060-b7219088", "recordingcheck|20071109-015844|1194573524.135408") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
[Kasterisk1*CLI>
recordingcheck|20071109-015844|1194573524.135408: No AMPUSER db entry for 9006. Not recording
[Kasterisk1*CLI>
-- AGI Script recordingcheck completed, returning 0
[Kasterisk1*CLI>
-- Executing NoOp("SIP/5060-b7219088", "No recording needed") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "0?skipoutcid") in new stack
-- Executing Set("SIP/5060-b7219088", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing Macro("SIP/5060-b7219088", "outbound-callerid|10") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing NoOp("SIP/5060-b7219088", "REALCALLERIDNUM is 9006") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing Set("SIP/5060-b7219088", "USEROUTCID=") in new stack
-- Executing Set("SIP/5060-b7219088", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/5060-b7219088", "TRUNKOUTCID=") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing GotoIf("SIP/5060-b7219088", "1?usercid") in new stack
-- Goto (macro-outbound-callerid,s,18)
-- Executing GotoIf("SIP/5060-b7219088", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing NoOp("SIP/5060-b7219088", "CallerID set to "" <9006>") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "0?nomax") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "0?chanfull") in new stack
-- Executing DeadAGI("SIP/5060-b7219088", "fixlocalprefix") in new stack
[Kasterisk1*CLI>
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
[Kasterisk1*CLI>
-- Executing Macro("SIP/5060-b722ed50", "dialout-trunk|10|9086265379||") in new stack
-- Executing Set("SIP/5060-b722ed50", "CDR(userfield)=204.14.9.50") in new stack
-- Executing Set("SIP/5060-b722ed50", "DIAL_TRUNK=10") in new stack
-- Executing Set("SIP/5060-b722ed50", "_NODEST=") in new stack
-- Executing Set("SIP/5060-b722ed50", "DIAL_NUMBER=9086265379") in new stack
-- Executing Set("SIP/5060-b722ed50", "ROUTE_PASSWD=") in new stack
-- Executing Set("SIP/5060-b722ed50", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing GotoIf("SIP/5060-b722ed50", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing Set("SIP/5060-b722ed50", "GROUP()=OUT_10") in new stack
-- Executing Macro("SIP/5060-b722ed50", "user-callerid|SKIPTTL") in new stack
-- Executing NoOp("SIP/5060-b722ed50", "user-callerid: 9006") in new stack
-- Executing GotoIf("SIP/5060-b722ed50", "0?report") in new stack
-- Executing GotoIf("SIP/5060-b722ed50", "0?start") in new stack
-- Executing Set("SIP/5060-b722ed50", "REALCALLERIDNUM=9006") in new stack
-- Executing NoOp("SIP/5060-b722ed50", "REALCALLERIDNUM is 9006") in new stack
-- Executing Set("SIP/5060-b722ed50", "AMPUSER=") in new stack
-- Executing Set("SIP/5060-b722ed50", "AMPUSERCIDNAME=") in new stack
-- Executing GotoIf("SIP/5060-b722ed50", "1?report") in new stack
-- Goto (macro-user-callerid,s,11)
-- Executing NoOp("SIP/5060-b722ed50", "TTL: ARG1: SKIPTTL") in new stack
-- Executing GotoIf("SIP/5060-b722ed50", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/5060-b722ed50", "Using CallerID "" <9006>") in new stack
-- Executing Macro("SIP/5060-b722ed50", "record-enable|9006|OUT") in new stack
-- Executing GotoIf("SIP/5060-b722ed50", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("SIP/5060-b722ed50", "recordingcheck|20071109-015845|1194573525.135409") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
[Kasterisk1*CLI>
recordingcheck|20071109-015845|1194573525.135409: No AMPUSER db entry for 9006. Not recording
[Kasterisk1*CLI>
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/5060-b722ed50", "No recording needed") in new stack
-- Executing GotoIf("SIP/5060-b722ed50", "0?skipoutcid") in new stack
-- Executing Set("SIP/5060-b722ed50", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing Macro("SIP/5060-b722ed50", "outbound-callerid|10") in new stack
-- Executing GotoIf("SIP/5060-b722ed50", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing NoOp("SIP/5060-b722ed50", "REALCALLERIDNUM is 9006") in new stack
-- Executing GotoIf("SIP/5060-b722ed50", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing Set("SIP/5060-b722ed50", "USEROUTCID=") in new stack
-- Executing Set("SIP/5060-b722ed50", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/5060-b722ed50", "TRUNKOUTCID=") in new stack
-- Executing GotoIf("SIP/5060-b722ed50", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing GotoIf("SIP/5060-b722ed50", "1?usercid") in new stack
-- Goto (macro-outbound-callerid,s,18)
-- Executing GotoIf("SIP/5060-b722ed50", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing NoOp("SIP/5060-b722ed50", "CallerID set to "" <9006>") in new stack
-- Executing GotoIf("SIP/5060-b722ed50", "0?nomax") in new stack
-- AGI Script fixlocalprefix completed, returning 0
-- Executing GotoIf("SIP/5060-b722ed50", "0?chanfull") in new stack
-- Executing DeadAGI("SIP/5060-b722ed50", "fixlocalprefix") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b7219088", "OUTNUM=9082655827") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
-- Executing Set("SIP/5060-b7219088", "custom=IAX2/GTT_No_VPN") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "0?customtrunk") in new stack
-- Executing Set("SIP/5060-b7219088", "CDR(userfield)=204.14.9.50") in new stack
-- Executing Dial("SIP/5060-b7219088", "IAX2/GTT_No_VPN/9082655827|300|") in new stack
-- Called GTT_No_VPN/9082655827
[Kasterisk1*CLI>
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/5060-b722ed50", "OUTNUM=9086265379") in new stack
-- Executing Set("SIP/5060-b722ed50", "custom=IAX2/GTT_No_VPN") in new stack
-- Executing GotoIf("SIP/5060-b722ed50", "0?customtrunk") in new stack
-- Executing Set("SIP/5060-b722ed50", "CDR(userfield)=204.14.9.50") in new stack
-- Executing Dial("SIP/5060-b722ed50", "IAX2/GTT_No_VPN/9086265379|300|") in new stack
-- Called GTT_No_VPN/9086265379
[Kasterisk1*CLI>
-- Call accepted by 190.80.53.168 (format g723)
-- Format for call is g723
[Kasterisk1*CLI>
-- Call accepted by 190.80.53.168 (format g723)
-- Format for call is g723
[Kasterisk1*CLI>
Sending to 64.192.196.12 : 5060 (non-NAT)
Found RTP audio format 18
Found RTP audio format 19
Peer audio RTP is at port 64.192.196.12:18578
Found description format G729
Found description format CN
Capabilities: us - 0x8070f (g723|gsm|ulaw|alaw|g729|speex|ilbc|h263), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x2 (CN), combined - 0x0 (nothing)
Looking for 18686243211 in from-sip-external (domain 208.75.208.90)
list_route: hop: <sip:64.192.196.12:5060>
Transmitting (no NAT) to 64.192.196.12:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 64.192.196.12:5060;branch=z9hG4bKF587;received=64.192.196.12
From: <sip:64.192.196.12>;tag=2A02E176-E4A
To: <sip:18686243211@208.75.208.90>
Call-ID: 2C1CA2B1-324711D6-98C6F52C-1F15DBCD@64.192.196.12
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:18686243211@208.75.208.90>
Content-Length: 0
---
-- Executing NoOp("SIP/64.192.196.12-b7229810", "Received incoming SIP connection from unknown peer to 18686243211") in new stack
-- Executing Set("SIP/64.192.196.12-b7229810", "DID=18686243211") in new stack
-- Executing Goto("SIP/64.192.196.12-b7229810", "s|1") in new stack
[Kasterisk1*CLI>
-- Goto (from-sip-external,s,1)
[Kasterisk1*CLI>
-- Executing Ringing("SIP/64.192.196.12-b7229810", "") in new stack
Transmitting (no NAT) to 64.192.196.12:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 64.192.196.12:5060;branch=z9hG4bKF587;received=64.192.196.12
From: <sip:64.192.196.12>;tag=2A02E176-E4A
To: <sip:18686243211@208.75.208.90>;tag=as307e0059
Call-ID: 2C1CA2B1-324711D6-98C6F52C-1F15DBCD@64.192.196.12
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:18686243211@208.75.208.90>
Content-Length: 0
---
-- Executing Set("SIP/64.192.196.12-b7229810", "CDR(userfield)=64.192.196.12") in new stack
-- Executing GotoIf("SIP/64.192.196.12-b7229810", "1?from-trunk|18686243211|1") in new stack
-- Goto (from-trunk,18686243211,1)
Scheduling destruction of call '2C1CA2B1-324711D6-98C6F52C-1F15DBCD@64.192.196.12' in 32000 ms
Reliably Transmitting (no NAT) to 64.192.196.12:5060:
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 64.192.196.12:5060;branch=z9hG4bKF587;received=64.192.196.12
From: <sip:64.192.196.12>;tag=2A02E176-E4A
To: <sip:18686243211@208.75.208.90>;tag=as307e0059
Call-ID: 2C1CA2B1-324711D6-98C6F52C-1F15DBCD@64.192.196.12
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:18686243211@208.75.208.90>
Content-Length: 0
---
[Kasterisk1*CLI>
<-- SIP read from 64.192.196.12:5060:
ACK sip:18686243211@208.75.208.90:5060 SIP/2.0
Via: SIP/2.0/UDP 64.192.196.12:5060;branch=z9hG4bKF587
From: <sip:64.192.196.12>;tag=2A02E176-E4A
To: <sip:18686243211@208.75.208.90>;tag=as307e0059
Date: Sat, 09 Mar 2002 03:47:11 GMT
Call-ID: 2C1CA2B1-324711D6-98C6F52C-1F15DBCD@64.192.196.12
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
--- (10 headers 0 lines) ---
[Kasterisk1*CLI>
-- Hungup 'IAX2/GTT_No_VPN-16409'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/64.15.135.252-b706d540' in macro 'dialout-trunk'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/64.15.135.252-b706d540'
-- Executing Macro("SIP/64.15.135.252-b706d540", "hangupcall") in new stack
-- Executing ResetCDR("SIP/64.15.135.252-b706d540", "w") in new stack
[Kasterisk1*CLI>
-- Executing NoCDR("SIP/64.15.135.252-b706d540", "") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b706d540", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing GotoIf("SIP/64.15.135.252-b706d540", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing Wait("SIP/64.15.135.252-b706d540", "5") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/64.15.135.252-b706d540' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/64.15.135.252-b706d540'
[Kasterisk1*CLI>
-- Executing Macro("SIP/5060-b700ed88", "dialout-trunk|10|9086260284||") in new stack
-- Executing Set("SIP/5060-b700ed88", "CDR(userfield)=70.42.39.67") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b700ed88", "DIAL_TRUNK=10") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b700ed88", "_NODEST=") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b700ed88", "DIAL_NUMBER=9086260284") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b700ed88", "ROUTE_PASSWD=") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b700ed88", "DIAL_TRUNK_OPTIONS=tr") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/5060-b700ed88", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing Set("SIP/5060-b700ed88", "GROUP()=OUT_10") in new stack
-- Executing Macro("SIP/5060-b700ed88", "user-callerid|SKIPTTL") in new stack
-- Executing NoOp("SIP/5060-b700ed88", "user-callerid: 9010") in new stack
-- Executing GotoIf("SIP/5060-b700ed88", "0?report") in new stack
-- Executing GotoIf("SIP/5060-b700ed88", "0?start") in new stack
-- Executing Set("SIP/5060-b700ed88", "REALCALLERIDNUM=9010") in new stack
-- Executing NoOp("SIP/5060-b700ed88", "REALCALLERIDNUM is 9010") in new stack
-- Executing Set("SIP/5060-b700ed88", "AMPUSER=") in new stack
-- Executing Set("SIP/5060-b700ed88", "AMPUSERCIDNAME=") in new stack
-- Executing GotoIf("SIP/5060-b700ed88", "1?report") in new stack
-- Goto (macro-user-callerid,s,11)
-- Executing NoOp("SIP/5060-b700ed88", "TTL: ARG1: SKIPTTL") in new stack
-- Executing GotoIf("SIP/5060-b700ed88", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/5060-b700ed88", "Using CallerID "" <9010>") in new stack
-- Executing Macro("SIP/5060-b700ed88", "record-enable|9010|OUT") in new stack
-- Executing GotoIf("SIP/5060-b700ed88", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("SIP/5060-b700ed88", "recordingcheck|20071109-015851|1194573531.135413") in new stack
[Kasterisk1*CLI>
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
[Kasterisk1*CLI>
recordingcheck|20071109-015851|1194573531.135413: No AMPUSER db entry for 9010. Not recording
[Kasterisk1*CLI>
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/5060-b700ed88", "No recording needed") in new stack
-- Executing GotoIf("SIP/5060-b700ed88", "0?skipoutcid") in new stack
-- Executing Set("SIP/5060-b700ed88", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing Macro("SIP/5060-b700ed88", "outbound-callerid|10") in new stack
-- Executing GotoIf("SIP/5060-b700ed88", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing NoOp("SIP/5060-b700ed88", "REALCALLERIDNUM is 9010") in new stack
-- Executing GotoIf("SIP/5060-b700ed88", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing Set("SIP/5060-b700ed88", "USEROUTCID=") in new stack
-- Executing Set("SIP/5060-b700ed88", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/5060-b700ed88", "TRUNKOUTCID=") in new stack
-- Executing GotoIf("SIP/5060-b700ed88", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing GotoIf("SIP/5060-b700ed88", "1?usercid") in new stack
-- Goto (macro-outbound-callerid,s,18)
-- Executing GotoIf("SIP/5060-b700ed88", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing NoOp("SIP/5060-b700ed88", "CallerID set to "" <9010>") in new stack
-- Executing GotoIf("SIP/5060-b700ed88", "0?nomax") in new stack
-- Executing GotoIf("SIP/5060-b700ed88", "0?chanfull") in new stack
-- Executing DeadAGI("SIP/5060-b700ed88", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
[Kasterisk1*CLI>
-- AGI Script fixlocalprefix completed, returning 0
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b700ed88", "OUTNUM=9086260284") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b700ed88", "custom=IAX2/GTT_No_VPN") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/5060-b700ed88", "0?customtrunk") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b700ed88", "CDR(userfield)=70.42.39.67") in new stack
[Kasterisk1*CLI>
-- Executing Dial("SIP/5060-b700ed88", "IAX2/GTT_No_VPN/9086260284|300|") in new stack
-- Called GTT_No_VPN/9086260284
[Kasterisk1*CLI>
-- Call accepted by 190.80.53.168 (format g723)
-- Format for call is g723
[Kasterisk1*CLI>
-- IAX2/Vydall-16399 stopped sounds
[Kasterisk1*CLI>
-- IAX2/Vydall-16399 is busy
-- Hungup 'IAX2/Vydall-16399'
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing Goto("SIP/64.15.135.252-b729f6f8", "s-BUSY|1") in new stack
-- Goto (macro-dialout-trunk,s-BUSY,1)
-- Executing NoOp("SIP/64.15.135.252-b729f6f8", "Dial failed due to trunk reporting BUSY - giving up") in new stack
-- Executing Set("SIP/64.15.135.252-b729f6f8", "CDR(userfield)=64.15.135.252") in new stack
-- Executing Macro("SIP/64.15.135.252-b729f6f8", "dialout-trunk|10|9086850715||") in new stack
-- Executing Set("SIP/64.15.135.252-b729f6f8", "CDR(userfield)=64.15.135.252") in new stack
-- Executing Set("SIP/64.15.135.252-b729f6f8", "DIAL_TRUNK=10") in new stack
-- Executing Set("SIP/64.15.135.252-b729f6f8", "_NODEST=") in new stack
-- Executing Set("SIP/64.15.135.252-b729f6f8", "DIAL_NUMBER=9086850715") in new stack
-- Executing Set("SIP/64.15.135.252-b729f6f8", "ROUTE_PASSWD=") in new stack
-- Executing Set("SIP/64.15.135.252-b729f6f8", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b729f6f8", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing Set("SIP/64.15.135.252-b729f6f8", "GROUP()=OUT_10") in new stack
-- Executing Macro("SIP/64.15.135.252-b729f6f8", "user-callerid|SKIPTTL") in new stack
-- Executing NoOp("SIP/64.15.135.252-b729f6f8", "user-callerid: 7aaa 9015") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b729f6f8", "0?report") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b729f6f8", "1?start") in new stack
-- Goto (macro-user-callerid,s,5)
-- Executing NoOp("SIP/64.15.135.252-b729f6f8", "REALCALLERIDNUM is 9015") in new stack
-- Executing Set("SIP/64.15.135.252-b729f6f8", "AMPUSER=") in new stack
-- Executing Set("SIP/64.15.135.252-b729f6f8", "AMPUSERCIDNAME=") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b729f6f8", "1?report") in new stack
-- Goto (macro-user-callerid,s,11)
[Kasterisk1*CLI>
-- Executing NoOp("SIP/64.15.135.252-b729f6f8", "TTL: ARG1: SKIPTTL") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b729f6f8", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/64.15.135.252-b729f6f8", "Using CallerID "7aaa" <9015>") in new stack
-- Executing Macro("SIP/64.15.135.252-b729f6f8", "record-enable|9015|OUT") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b729f6f8", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("SIP/64.15.135.252-b729f6f8", "recordingcheck|20071109-015854|1194573518.135400") in new stack
[Kasterisk1*CLI>
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
[Kasterisk1*CLI>
recordingcheck|20071109-015854|1194573518.135400: No AMPUSER db entry for 9015. Not recording
[Kasterisk1*CLI>
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/64.15.135.252-b729f6f8", "No recording needed") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b729f6f8", "0?skipoutcid") in new stack
-- Executing Set("SIP/64.15.135.252-b729f6f8", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing Macro("SIP/64.15.135.252-b729f6f8", "outbound-callerid|10") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b729f6f8", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing NoOp("SIP/64.15.135.252-b729f6f8", "REALCALLERIDNUM is 9015") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b729f6f8", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing Set("SIP/64.15.135.252-b729f6f8", "USEROUTCID=") in new stack
-- Executing Set("SIP/64.15.135.252-b729f6f8", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/64.15.135.252-b729f6f8", "TRUNKOUTCID=") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b729f6f8", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing GotoIf("SIP/64.15.135.252-b729f6f8", "1?usercid") in new stack
-- Goto (macro-outbound-callerid,s,18)
-- Executing GotoIf("SIP/64.15.135.252-b729f6f8", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing NoOp("SIP/64.15.135.252-b729f6f8", "CallerID set to "7aaa" <9015>") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b729f6f8", "0?nomax") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b729f6f8", "0?chanfull") in new stack
-- Executing DeadAGI("SIP/64.15.135.252-b729f6f8", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
[Kasterisk1*CLI>
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/64.15.135.252-b729f6f8", "OUTNUM=9086850715") in new stack
-- Executing Set("SIP/64.15.135.252-b729f6f8", "custom=IAX2/GTT_No_VPN") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b729f6f8", "0?customtrunk") in new stack
-- Executing Set("SIP/64.15.135.252-b729f6f8", "CDR(userfield)=64.15.135.252") in new stack
-- Executing Dial("SIP/64.15.135.252-b729f6f8", "IAX2/GTT_No_VPN/9086850715|300|") in new stack
-- Called GTT_No_VPN/9086850715
[Kasterisk1*CLI>
-- Call accepted by 190.80.53.168 (format g723)
-- Format for call is g723
[Kasterisk1*CLI>
-- Hungup 'IAX2/GTT_No_VPN-16411'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/5060-b700ed88' in macro 'dialout-trunk'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/5060-b700ed88'
-- Executing Macro("SIP/5060-b700ed88", "hangupcall") in new stack
-- Executing ResetCDR("SIP/5060-b700ed88", "w") in new stack
[Kasterisk1*CLI>
-- Executing NoCDR("SIP/5060-b700ed88", "") in new stack
-- Executing GotoIf("SIP/5060-b700ed88", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing GotoIf("SIP/5060-b700ed88", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing Wait("SIP/5060-b700ed88", "5") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/5060-b700ed88' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/5060-b700ed88'
[Kasterisk1*CLI>
-- Hungup 'IAX2/GTT_No_VPN-16393'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/204.14.9.50-b720cfe0' in macro 'dialout-trunk'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/204.14.9.50-b720cfe0'
-- Executing Macro("SIP/204.14.9.50-b720cfe0", "hangupcall") in new stack
-- Executing ResetCDR("SIP/204.14.9.50-b720cfe0", "w") in new stack
[Kasterisk1*CLI>
-- Executing NoCDR("SIP/204.14.9.50-b720cfe0", "") in new stack
-- Executing GotoIf("SIP/204.14.9.50-b720cfe0", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing GotoIf("SIP/204.14.9.50-b720cfe0", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing Wait("SIP/204.14.9.50-b720cfe0", "5") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/204.14.9.50-b720cfe0' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/204.14.9.50-b720cfe0'
[Kasterisk1*CLI>
-- Executing Macro("SIP/5060-b720cfe0", "dialout-trunk|10|9086435737||") in new stack
-- Executing Set("SIP/5060-b720cfe0", "CDR(userfield)=204.14.9.50") in new stack
-- Executing Set("SIP/5060-b720cfe0", "DIAL_TRUNK=10") in new stack
-- Executing Set("SIP/5060-b720cfe0", "_NODEST=") in new stack
-- Executing Set("SIP/5060-b720cfe0", "DIAL_NUMBER=9086435737") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b720cfe0", "ROUTE_PASSWD=") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b720cfe0", "DIAL_TRUNK_OPTIONS=tr") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/5060-b720cfe0", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing Set("SIP/5060-b720cfe0", "GROUP()=OUT_10") in new stack
-- Executing Macro("SIP/5060-b720cfe0", "user-callerid|SKIPTTL") in new stack
[Kasterisk1*CLI>
-- Executing NoOp("SIP/5060-b720cfe0", "user-callerid: 9006") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/5060-b720cfe0", "0?report") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/5060-b720cfe0", "0?start") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b720cfe0", "REALCALLERIDNUM=9006") in new stack
[Kasterisk1*CLI>
-- Executing NoOp("SIP/5060-b720cfe0", "REALCALLERIDNUM is 9006") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b720cfe0", "AMPUSER=") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b720cfe0", "AMPUSERCIDNAME=") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/5060-b720cfe0", "1?report") in new stack
-- Goto (macro-user-callerid,s,11)
-- Executing NoOp("SIP/5060-b720cfe0", "TTL: ARG1: SKIPTTL") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/5060-b720cfe0", "1?continue") in new stack
[Kasterisk1*CLI>
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/5060-b720cfe0", "Using CallerID "" <9006>") in new stack
[Kasterisk1*CLI>
-- Executing Macro("SIP/5060-b720cfe0", "record-enable|9006|OUT") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/5060-b720cfe0", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
[Kasterisk1*CLI>
-- Executing DeadAGI("SIP/5060-b720cfe0", "recordingcheck|20071109-015856|1194573536.135416") in new stack
[Kasterisk1*CLI>
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
[Kasterisk1*CLI>
recordingcheck|20071109-015856|1194573536.135416: No AMPUSER db entry for 9006. Not recording
[Kasterisk1*CLI>
-- AGI Script recordingcheck completed, returning 0
[Kasterisk1*CLI>
-- Executing NoOp("SIP/5060-b720cfe0", "No recording needed") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/5060-b720cfe0", "0?skipoutcid") in new stack
-- Executing Set("SIP/5060-b720cfe0", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing Macro("SIP/5060-b720cfe0", "outbound-callerid|10") in new stack
-- Executing GotoIf("SIP/5060-b720cfe0", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
[Kasterisk1*CLI>
-- Executing NoOp("SIP/5060-b720cfe0", "REALCALLERIDNUM is 9006") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/5060-b720cfe0", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing Set("SIP/5060-b720cfe0", "USEROUTCID=") in new stack
-- Executing Set("SIP/5060-b720cfe0", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/5060-b720cfe0", "TRUNKOUTCID=") in new stack
-- Executing GotoIf("SIP/5060-b720cfe0", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing GotoIf("SIP/5060-b720cfe0", "1?usercid") in new stack
-- Goto (macro-outbound-callerid,s,18)
-- Executing GotoIf("SIP/5060-b720cfe0", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing NoOp("SIP/5060-b720cfe0", "CallerID set to "" <9006>") in new stack
-- Executing GotoIf("SIP/5060-b720cfe0", "0?nomax") in new stack
-- Executing GotoIf("SIP/5060-b720cfe0", "0?chanfull") in new stack
[Kasterisk1*CLI>
-- Executing DeadAGI("SIP/5060-b720cfe0", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
[Kasterisk1*CLI>
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/5060-b720cfe0", "OUTNUM=9086435737") in new stack
-- Executing Set("SIP/5060-b720cfe0", "custom=IAX2/GTT_No_VPN") in new stack
-- Executing GotoIf("SIP/5060-b720cfe0", "0?customtrunk") in new stack
-- Executing Set("SIP/5060-b720cfe0", "CDR(userfield)=204.14.9.50") in new stack
-- Executing Dial("SIP/5060-b720cfe0", "IAX2/GTT_No_VPN/9086435737|300|") in new stack
-- Called GTT_No_VPN/9086435737
[Kasterisk1*CLI>
-- Call accepted by 190.80.53.168 (format g723)
-- Format for call is g723
[Kasterisk1*CLI>
-- Hungup 'IAX2/GTT_No_VPN-16432'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/66.246.197.20-b721c238' in macro 'dialout-trunk'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/66.246.197.20-b721c238'
-- Executing Macro("SIP/66.246.197.20-b721c238", "hangupcall") in new stack
[Kasterisk1*CLI>
-- Executing ResetCDR("SIP/66.246.197.20-b721c238", "w") in new stack
[Kasterisk1*CLI>
-- Executing NoCDR("SIP/66.246.197.20-b721c238", "") in new stack
-- Executing GotoIf("SIP/66.246.197.20-b721c238", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing GotoIf("SIP/66.246.197.20-b721c238", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing Wait("SIP/66.246.197.20-b721c238", "5") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/66.246.197.20-b721c238' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/66.246.197.20-b721c238'
== End MixMonitor Recording SIP/66.246.197.20-b721c238
[Kasterisk1*CLI>
-- IAX2/GTT_No_VPN-16407 is ringing
[Kasterisk1*CLI>
-- Executing Macro("SIP/5060-b70a51d8", "dialout-trunk|10|9082266036||") in new stack
-- Executing Set("SIP/5060-b70a51d8", "CDR(userfield)=70.42.39.67") in new stack
-- Executing Set("SIP/5060-b70a51d8", "DIAL_TRUNK=10") in new stack
-- Executing Set("SIP/5060-b70a51d8", "_NODEST=") in new stack
-- Executing Set("SIP/5060-b70a51d8", "DIAL_NUMBER=9082266036") in new stack
-- Executing Set("SIP/5060-b70a51d8", "ROUTE_PASSWD=") in new stack
-- Executing Set("SIP/5060-b70a51d8", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing GotoIf("SIP/5060-b70a51d8", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing Set("SIP/5060-b70a51d8", "GROUP()=OUT_10") in new stack
-- Executing Macro("SIP/5060-b70a51d8", "user-callerid|SKIPTTL") in new stack
-- Executing NoOp("SIP/5060-b70a51d8", "user-callerid: 9010") in new stack
-- Executing GotoIf("SIP/5060-b70a51d8", "0?report") in new stack
-- Executing GotoIf("SIP/5060-b70a51d8", "0?start") in new stack
-- Executing Set("SIP/5060-b70a51d8", "REALCALLERIDNUM=9010") in new stack
-- Executing NoOp("SIP/5060-b70a51d8", "REALCALLERIDNUM is 9010") in new stack
-- Executing Set("SIP/5060-b70a51d8", "AMPUSER=") in new stack
-- Executing Set("SIP/5060-b70a51d8", "AMPUSERCIDNAME=") in new stack
-- Executing GotoIf("SIP/5060-b70a51d8", "1?report") in new stack
-- Goto (macro-user-callerid,s,11)
-- Executing NoOp("SIP/5060-b70a51d8", "TTL: ARG1: SKIPTTL") in new stack
-- Executing GotoIf("SIP/5060-b70a51d8", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/5060-b70a51d8", "Using CallerID "" <9010>") in new stack
-- Executing Macro("SIP/5060-b70a51d8", "record-enable|9010|OUT") in new stack
-- Executing GotoIf("SIP/5060-b70a51d8", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("SIP/5060-b70a51d8", "recordingcheck|20071109-015859|1194573539.135418") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
[Kasterisk1*CLI>
recordingcheck|20071109-015859|1194573539.135418: No AMPUSER db entry for 9010. Not recording
[Kasterisk1*CLI>
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/5060-b70a51d8", "No recording needed") in new stack
-- Executing GotoIf("SIP/5060-b70a51d8", "0?skipoutcid") in new stack
-- Executing Set("SIP/5060-b70a51d8", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing Macro("SIP/5060-b70a51d8", "outbound-callerid|10") in new stack
-- Executing GotoIf("SIP/5060-b70a51d8", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing NoOp("SIP/5060-b70a51d8", "REALCALLERIDNUM is 9010") in new stack
-- Executing GotoIf("SIP/5060-b70a51d8", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing Set("SIP/5060-b70a51d8", "USEROUTCID=") in new stack
-- Executing Set("SIP/5060-b70a51d8", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/5060-b70a51d8", "TRUNKOUTCID=") in new stack
-- Executing GotoIf("SIP/5060-b70a51d8", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing GotoIf("SIP/5060-b70a51d8", "1?usercid") in new stack
-- Goto (macro-outbound-callerid,s,18)
-- Executing GotoIf("SIP/5060-b70a51d8", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing NoOp("SIP/5060-b70a51d8", "CallerID set to "" <9010>") in new stack
-- Executing GotoIf("SIP/5060-b70a51d8", "0?nomax") in new stack
-- Executing GotoIf("SIP/5060-b70a51d8", "0?chanfull") in new stack
-- Executing DeadAGI("SIP/5060-b70a51d8", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
[Kasterisk1*CLI>
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/5060-b70a51d8", "OUTNUM=9082266036") in new stack
-- Executing Set("SIP/5060-b70a51d8", "custom=IAX2/GTT_No_VPN") in new stack
-- Executing GotoIf("SIP/5060-b70a51d8", "0?customtrunk") in new stack
-- Executing Set("SIP/5060-b70a51d8", "CDR(userfield)=70.42.39.67") in new stack
-- Executing Dial("SIP/5060-b70a51d8", "IAX2/GTT_No_VPN/9082266036|300|") in new stack
-- Called GTT_No_VPN/9082266036
[Kasterisk1*CLI>
-- Call accepted by 190.80.53.168 (format g723)
-- Format for call is g723
[Kasterisk1*CLI>
-- IAX2/GTT_No_VPN-16387 stopped sounds
[Kasterisk1*CLI>
-- IAX2/GTT_No_VPN-16387 answered SIP/5060-b72f6b18
[Kasterisk1*CLI>
-- IAX2/GTT_No_VPN-16416 is ringing
[Kasterisk1*CLI>
-- Executing Macro("SIP/64.15.135.252-b72e2d80", "dialout-trunk|10|9082183537||") in new stack
-- Executing Set("SIP/64.15.135.252-b72e2d80", "CDR(userfield)=64.15.135.252") in new stack
-- Executing Set("SIP/64.15.135.252-b72e2d80", "DIAL_TRUNK=10") in new stack
-- Executing Set("SIP/64.15.135.252-b72e2d80", "_NODEST=") in new stack
-- Executing Set("SIP/64.15.135.252-b72e2d80", "DIAL_NUMBER=9082183537") in new stack
-- Executing Set("SIP/64.15.135.252-b72e2d80", "ROUTE_PASSWD=") in new stack
-- Executing Set("SIP/64.15.135.252-b72e2d80", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b72e2d80", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing Set("SIP/64.15.135.252-b72e2d80", "GROUP()=OUT_10") in new stack
-- Executing Macro("SIP/64.15.135.252-b72e2d80", "user-callerid|SKIPTTL") in new stack
-- Executing NoOp("SIP/64.15.135.252-b72e2d80", "user-callerid: 7aaa 9015") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b72e2d80", "0?report") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b72e2d80", "0?start") in new stack
-- Executing Set("SIP/64.15.135.252-b72e2d80", "REALCALLERIDNUM=9015") in new stack
-- Executing NoOp("SIP/64.15.135.252-b72e2d80", "REALCALLERIDNUM is 9015") in new stack
-- Executing Set("SIP/64.15.135.252-b72e2d80", "AMPUSER=") in new stack
-- Executing Set("SIP/64.15.135.252-b72e2d80", "AMPUSERCIDNAME=") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b72e2d80", "1?report") in new stack
-- Goto (macro-user-callerid,s,11)
-- Executing NoOp("SIP/64.15.135.252-b72e2d80", "TTL: ARG1: SKIPTTL") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b72e2d80", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/64.15.135.252-b72e2d80", "Using CallerID "7aaa" <9015>") in new stack
-- Executing Macro("SIP/64.15.135.252-b72e2d80", "record-enable|9015|OUT") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b72e2d80", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("SIP/64.15.135.252-b72e2d80", "recordingcheck|20071109-015902|1194573542.135420") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
[Kasterisk1*CLI>
recordingcheck|20071109-015902|1194573542.135420: No AMPUSER db entry for 9015. Not recording
[Kasterisk1*CLI>
-- AGI Script recordingcheck completed, returning 0
[Kasterisk1*CLI>
-- Executing NoOp("SIP/64.15.135.252-b72e2d80", "No recording needed") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/64.15.135.252-b72e2d80", "0?skipoutcid") in new stack
-- Executing Set("SIP/64.15.135.252-b72e2d80", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing Macro("SIP/64.15.135.252-b72e2d80", "outbound-callerid|10") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b72e2d80", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing NoOp("SIP/64.15.135.252-b72e2d80", "REALCALLERIDNUM is 9015") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b72e2d80", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
[Kasterisk1*CLI>
-- Executing Set("SIP/64.15.135.252-b72e2d80", "USEROUTCID=") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/64.15.135.252-b72e2d80", "EMERGENCYCID=") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/64.15.135.252-b72e2d80", "TRUNKOUTCID=") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/64.15.135.252-b72e2d80", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing GotoIf("SIP/64.15.135.252-b72e2d80", "1?usercid") in new stack
-- Goto (macro-outbound-callerid,s,18)
-- Executing GotoIf("SIP/64.15.135.252-b72e2d80", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
[Kasterisk1*CLI>
-- Executing NoOp("SIP/64.15.135.252-b72e2d80", "CallerID set to "7aaa" <9015>") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/64.15.135.252-b72e2d80", "0?nomax") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/64.15.135.252-b72e2d80", "0?chanfull") in new stack
[Kasterisk1*CLI>
-- Executing DeadAGI("SIP/64.15.135.252-b72e2d80", "fixlocalprefix") in new stack
[Kasterisk1*CLI>
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
[Kasterisk1*CLI>
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/64.15.135.252-b72e2d80", "OUTNUM=9082183537") in new stack
-- Executing Set("SIP/64.15.135.252-b72e2d80", "custom=IAX2/GTT_No_VPN") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b72e2d80", "0?customtrunk") in new stack
-- Executing Set("SIP/64.15.135.252-b72e2d80", "CDR(userfield)=64.15.135.252") in new stack
-- Executing Dial("SIP/64.15.135.252-b72e2d80", "IAX2/GTT_No_VPN/9082183537|300|") in new stack
-- Called GTT_No_VPN/9082183537
[Kasterisk1*CLI>
-- Call accepted by 190.80.53.168 (format g723)
-- Format for call is g723
[Kasterisk1*CLI>
-- Hungup 'IAX2/GTT_No_VPN-16403'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/5060-b7243938' in macro 'dialout-trunk'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/5060-b7243938'
-- Executing Macro("SIP/5060-b7243938", "hangupcall") in new stack
-- Executing ResetCDR("SIP/5060-b7243938", "w") in new stack
[Kasterisk1*CLI>
-- Executing NoCDR("SIP/5060-b7243938", "") in new stack
-- Executing GotoIf("SIP/5060-b7243938", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing GotoIf("SIP/5060-b7243938", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing Wait("SIP/5060-b7243938", "5") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/5060-b7243938' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/5060-b7243938'
[Kasterisk1*CLI>
-- Hungup 'IAX2/GTT_No_VPN-16391'
[Kasterisk1*CLI>
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/5060-b729c010' in macro 'dialout-trunk'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/5060-b729c010'
-- Executing Macro("SIP/5060-b729c010", "hangupcall") in new stack
-- Executing ResetCDR("SIP/5060-b729c010", "w") in new stack
[Kasterisk1*CLI>
-- Executing NoCDR("SIP/5060-b729c010", "") in new stack
-- Executing GotoIf("SIP/5060-b729c010", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing GotoIf("SIP/5060-b729c010", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing Wait("SIP/5060-b729c010", "5") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/5060-b729c010' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/5060-b729c010'
[Kasterisk1*CLI>
-- Hungup 'IAX2/GTT_No_VPN-16385'
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Goto("SIP/5060-b7219088", "s-CHANUNAVAIL|1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing NoOp("SIP/5060-b7219088", "Dial failed due to CHANUNAVAIL - failing through to other trunks") in new stack
-- Executing Macro("SIP/5060-b7219088", "dialout-trunk|5|9082655827||") in new stack
-- Executing Set("SIP/5060-b7219088", "CDR(userfield)=204.14.9.50") in new stack
-- Executing Set("SIP/5060-b7219088", "DIAL_TRUNK=5") in new stack
-- Executing Set("SIP/5060-b7219088", "_NODEST=") in new stack
-- Executing Set("SIP/5060-b7219088", "DIAL_NUMBER=9082655827") in new stack
-- Executing Set("SIP/5060-b7219088", "ROUTE_PASSWD=") in new stack
-- Executing Set("SIP/5060-b7219088", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing Set("SIP/5060-b7219088", "GROUP()=OUT_5") in new stack
-- Executing Macro("SIP/5060-b7219088", "user-callerid|SKIPTTL") in new stack
-- Executing NoOp("SIP/5060-b7219088", "user-callerid: 9006") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "0?report") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "1?start") in new stack
-- Goto (macro-user-callerid,s,5)
-- Executing NoOp("SIP/5060-b7219088", "REALCALLERIDNUM is 9006") in new stack
-- Executing Set("SIP/5060-b7219088", "AMPUSER=") in new stack
-- Executing Set("SIP/5060-b7219088", "AMPUSERCIDNAME=") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "1?report") in new stack
-- Goto (macro-user-callerid,s,11)
-- Executing NoOp("SIP/5060-b7219088", "TTL: ARG1: SKIPTTL") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/5060-b7219088", "Using CallerID "" <9006>") in new stack
-- Executing Macro("SIP/5060-b7219088", "record-enable|9006|OUT") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("SIP/5060-b7219088", "recordingcheck|20071109-015905|1194573524.135408") in new stack
[Kasterisk1*CLI>
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
[Kasterisk1*CLI>
recordingcheck|20071109-015905|1194573524.135408: No AMPUSER db entry for 9006. Not recording
[Kasterisk1*CLI>
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/5060-b7219088", "No recording needed") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "0?skipoutcid") in new stack
-- Executing Set("SIP/5060-b7219088", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing Macro("SIP/5060-b7219088", "outbound-callerid|5") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing NoOp("SIP/5060-b7219088", "REALCALLERIDNUM is 9006") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing Set("SIP/5060-b7219088", "USEROUTCID=") in new stack
-- Executing Set("SIP/5060-b7219088", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/5060-b7219088", "TRUNKOUTCID=") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing GotoIf("SIP/5060-b7219088", "1?usercid") in new stack
-- Goto (macro-outbound-callerid,s,18)
-- Executing GotoIf("SIP/5060-b7219088", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing NoOp("SIP/5060-b7219088", "CallerID set to "" <9006>") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "0?nomax") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "0?chanfull") in new stack
-- Executing DeadAGI("SIP/5060-b7219088", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
[Kasterisk1*CLI>
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/5060-b7219088", "OUTNUM=9082655827") in new stack
-- Executing Set("SIP/5060-b7219088", "custom=IAX2/VPN") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "0?customtrunk") in new stack
-- Executing Set("SIP/5060-b7219088", "CDR(userfield)=204.14.9.50") in new stack
-- Executing Dial("SIP/5060-b7219088", "IAX2/VPN/9082655827|300|") in new stack
== Everyone is busy/congested at this time (1:0/0/1)
[Kasterisk1*CLI>
-- Executing Goto("SIP/5060-b7219088", "s-CHANUNAVAIL|1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing NoOp("SIP/5060-b7219088", "Dial failed due to CHANUNAVAIL - failing through to other trunks") in new stack
-- Executing Macro("SIP/5060-b7219088", "dialout-trunk|17|9082655827||") in new stack
-- Executing Set("SIP/5060-b7219088", "CDR(userfield)=204.14.9.50") in new stack
-- Executing Set("SIP/5060-b7219088", "DIAL_TRUNK=17") in new stack
-- Executing Set("SIP/5060-b7219088", "_NODEST=") in new stack
-- Executing Set("SIP/5060-b7219088", "DIAL_NUMBER=9082655827") in new stack
-- Executing Set("SIP/5060-b7219088", "ROUTE_PASSWD=") in new stack
-- Executing Set("SIP/5060-b7219088", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing Set("SIP/5060-b7219088", "GROUP()=OUT_17") in new stack
-- Executing Macro("SIP/5060-b7219088", "user-callerid|SKIPTTL") in new stack
-- Executing NoOp("SIP/5060-b7219088", "user-callerid: 9006") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "0?report") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "1?start") in new stack
-- Goto (macro-user-callerid,s,5)
-- Executing NoOp("SIP/5060-b7219088", "REALCALLERIDNUM is 9006") in new stack
-- Executing Set("SIP/5060-b7219088", "AMPUSER=") in new stack
-- Executing Set("SIP/5060-b7219088", "AMPUSERCIDNAME=") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "1?report") in new stack
-- Goto (macro-user-callerid,s,11)
-- Executing NoOp("SIP/5060-b7219088", "TTL: ARG1: SKIPTTL") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/5060-b7219088", "Using CallerID "" <9006>") in new stack
-- Executing Macro("SIP/5060-b7219088", "record-enable|9006|OUT") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("SIP/5060-b7219088", "recordingcheck|20071109-015905|1194573524.135408") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
[Kasterisk1*CLI>
recordingcheck|20071109-015905|1194573524.135408: No AMPUSER db entry for 9006. Not recording
[Kasterisk1*CLI>
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/5060-b7219088", "No recording needed") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "0?skipoutcid") in new stack
-- Executing Set("SIP/5060-b7219088", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing Macro("SIP/5060-b7219088", "outbound-callerid|17") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing NoOp("SIP/5060-b7219088", "REALCALLERIDNUM is 9006") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing Set("SIP/5060-b7219088", "USEROUTCID=") in new stack
-- Executing Set("SIP/5060-b7219088", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/5060-b7219088", "TRUNKOUTCID=") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing GotoIf("SIP/5060-b7219088", "1?usercid") in new stack
-- Goto (macro-outbound-callerid,s,18)
-- Executing GotoIf("SIP/5060-b7219088", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing NoOp("SIP/5060-b7219088", "CallerID set to "" <9006>") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "0?nomax") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "0?chanfull") in new stack
-- Executing DeadAGI("SIP/5060-b7219088", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
[Kasterisk1*CLI>
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/5060-b7219088", "OUTNUM=9082655827") in new stack
-- Executing Set("SIP/5060-b7219088", "custom=IAX2/Vydall") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "0?customtrunk") in new stack
-- Executing Set("SIP/5060-b7219088", "CDR(userfield)=204.14.9.50") in new stack
-- Executing Dial("SIP/5060-b7219088", "IAX2/Vydall/9082655827|300|") in new stack
-- Called Vydall/9082655827
[Kasterisk1*CLI>
-- Call accepted by 190.80.53.205 (format g723)
-- Format for call is g723
[Kasterisk1*CLI>
-- Hungup 'IAX2/GTT_No_VPN-16405'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/5060-b722ed50' in macro 'dialout-trunk'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/5060-b722ed50'
-- Executing Macro("SIP/5060-b722ed50", "hangupcall") in new stack
-- Executing ResetCDR("SIP/5060-b722ed50", "w") in new stack
[Kasterisk1*CLI>
-- Executing NoCDR("SIP/5060-b722ed50", "") in new stack
-- Executing GotoIf("SIP/5060-b722ed50", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing GotoIf("SIP/5060-b722ed50", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing Wait("SIP/5060-b722ed50", "5") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/5060-b722ed50' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/5060-b722ed50'
[Kasterisk1*CLI>
-- Hungup 'IAX2/GTT_No_VPN-16413'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/5060-b7212548' in macro 'dialout-trunk'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/5060-b7212548'
-- Executing Macro("SIP/5060-b7212548", "hangupcall") in new stack
-- Executing ResetCDR("SIP/5060-b7212548", "w") in new stack
[Kasterisk1*CLI>
-- Executing NoCDR("SIP/5060-b7212548", "") in new stack
-- Executing GotoIf("SIP/5060-b7212548", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing GotoIf("SIP/5060-b7212548", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing Wait("SIP/5060-b7212548", "5") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/5060-b7212548' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/5060-b7212548'
[Kasterisk1*CLI>
-- Executing Macro("SIP/64.15.135.252-b722ed50", "dialout-trunk|10|9082258520||") in new stack
-- Executing Set("SIP/64.15.135.252-b722ed50", "CDR(userfield)=64.15.135.252") in new stack
-- Executing Set("SIP/64.15.135.252-b722ed50", "DIAL_TRUNK=10") in new stack
-- Executing Set("SIP/64.15.135.252-b722ed50", "_NODEST=") in new stack
-- Executing Set("SIP/64.15.135.252-b722ed50", "DIAL_NUMBER=9082258520") in new stack
-- Executing Set("SIP/64.15.135.252-b722ed50", "ROUTE_PASSWD=") in new stack
-- Executing Set("SIP/64.15.135.252-b722ed50", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b722ed50", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing Set("SIP/64.15.135.252-b722ed50", "GROUP()=OUT_10") in new stack
-- Executing Macro("SIP/64.15.135.252-b722ed50", "user-callerid|SKIPTTL") in new stack
-- Executing NoOp("SIP/64.15.135.252-b722ed50", "user-callerid: 7aaa 9015") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b722ed50", "0?report") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b722ed50", "0?start") in new stack
-- Executing Set("SIP/64.15.135.252-b722ed50", "REALCALLERIDNUM=9015") in new stack
-- Executing NoOp("SIP/64.15.135.252-b722ed50", "REALCALLERIDNUM is 9015") in new stack
-- Executing Set("SIP/64.15.135.252-b722ed50", "AMPUSER=") in new stack
-- Executing Set("SIP/64.15.135.252-b722ed50", "AMPUSERCIDNAME=") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b722ed50", "1?report") in new stack
-- Goto (macro-user-callerid,s,11)
-- Executing NoOp("SIP/64.15.135.252-b722ed50", "TTL: ARG1: SKIPTTL") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b722ed50", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/64.15.135.252-b722ed50", "Using CallerID "7aaa" <9015>") in new stack
-- Executing Macro("SIP/64.15.135.252-b722ed50", "record-enable|9015|OUT") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b722ed50", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("SIP/64.15.135.252-b722ed50", "recordingcheck|20071109-015908|1194573548.135423") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
[Kasterisk1*CLI>
recordingcheck|20071109-015908|1194573548.135423: No AMPUSER db entry for 9015. Not recording
[Kasterisk1*CLI>
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/64.15.135.252-b722ed50", "No recording needed") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/64.15.135.252-b722ed50", "0?skipoutcid") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/64.15.135.252-b722ed50", "DIAL_TRUNK_OPTIONS=") in new stack
[Kasterisk1*CLI>
-- Executing Macro("SIP/64.15.135.252-b722ed50", "outbound-callerid|10") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/64.15.135.252-b722ed50", "1?start") in new stack
[Kasterisk1*CLI>
-- Goto (macro-outbound-callerid,s,3)
[Kasterisk1*CLI>
-- Executing NoOp("SIP/64.15.135.252-b722ed50", "REALCALLERIDNUM is 9015") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/64.15.135.252-b722ed50", "1?normcid") in new stack
[Kasterisk1*CLI>
-- Goto (macro-outbound-callerid,s,9)
[Kasterisk1*CLI>
-- Executing Set("SIP/64.15.135.252-b722ed50", "USEROUTCID=") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/64.15.135.252-b722ed50", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/64.15.135.252-b722ed50", "TRUNKOUTCID=") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b722ed50", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/64.15.135.252-b722ed50", "1?usercid") in new stack
-- Goto (macro-outbound-callerid,s,18)
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/64.15.135.252-b722ed50", "1?report") in new stack
[Kasterisk1*CLI>
-- Goto (macro-outbound-callerid,s,22)
[Kasterisk1*CLI>
-- Executing NoOp("SIP/64.15.135.252-b722ed50", "CallerID set to "7aaa" <9015>") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/64.15.135.252-b722ed50", "0?nomax") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/64.15.135.252-b722ed50", "0?chanfull") in new stack
[Kasterisk1*CLI>
-- Executing DeadAGI("SIP/64.15.135.252-b722ed50", "fixlocalprefix") in new stack
[Kasterisk1*CLI>
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
[Kasterisk1*CLI>
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/64.15.135.252-b722ed50", "OUTNUM=9082258520") in new stack
-- Executing Set("SIP/64.15.135.252-b722ed50", "custom=IAX2/GTT_No_VPN") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b722ed50", "0?customtrunk") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/64.15.135.252-b722ed50", "CDR(userfield)=64.15.135.252") in new stack
-- Executing Dial("SIP/64.15.135.252-b722ed50", "IAX2/GTT_No_VPN/9082258520|300|") in new stack
[Kasterisk1*CLI>
-- Called GTT_No_VPN/9082258520
[Kasterisk1*CLI>
-- Call accepted by 190.80.53.168 (format g723)
-- Format for call is g723
[Kasterisk1*CLI>
-- Executing Macro("SIP/5060-b72a8b70", "dialout-trunk|10|9086455009||") in new stack
-- Executing Set("SIP/5060-b72a8b70", "CDR(userfield)=204.14.9.50") in new stack
-- Executing Set("SIP/5060-b72a8b70", "DIAL_TRUNK=10") in new stack
-- Executing Set("SIP/5060-b72a8b70", "_NODEST=") in new stack
-- Executing Set("SIP/5060-b72a8b70", "DIAL_NUMBER=9086455009") in new stack
-- Executing Set("SIP/5060-b72a8b70", "ROUTE_PASSWD=") in new stack
-- Executing Set("SIP/5060-b72a8b70", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing GotoIf("SIP/5060-b72a8b70", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing Set("SIP/5060-b72a8b70", "GROUP()=OUT_10") in new stack
-- Executing Macro("SIP/5060-b72a8b70", "user-callerid|SKIPTTL") in new stack
-- Executing NoOp("SIP/5060-b72a8b70", "user-callerid: 9006") in new stack
-- Executing GotoIf("SIP/5060-b72a8b70", "0?report") in new stack
-- Executing GotoIf("SIP/5060-b72a8b70", "0?start") in new stack
-- Executing Set("SIP/5060-b72a8b70", "REALCALLERIDNUM=9006") in new stack
-- Executing NoOp("SIP/5060-b72a8b70", "REALCALLERIDNUM is 9006") in new stack
-- Executing Set("SIP/5060-b72a8b70", "AMPUSER=") in new stack
-- Executing Set("SIP/5060-b72a8b70", "AMPUSERCIDNAME=") in new stack
-- Executing GotoIf("SIP/5060-b72a8b70", "1?report") in new stack
-- Goto (macro-user-callerid,s,11)
-- Executing NoOp("SIP/5060-b72a8b70", "TTL: ARG1: SKIPTTL") in new stack
-- Executing GotoIf("SIP/5060-b72a8b70", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/5060-b72a8b70", "Using CallerID "" <9006>") in new stack
-- Executing Macro("SIP/5060-b72a8b70", "record-enable|9006|OUT") in new stack
-- Executing GotoIf("SIP/5060-b72a8b70", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("SIP/5060-b72a8b70", "recordingcheck|20071109-015909|1194573549.135425") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
[Kasterisk1*CLI>
recordingcheck|20071109-015909|1194573549.135425: No AMPUSER db entry for 9006. Not recording
[Kasterisk1*CLI>
-- AGI Script recordingcheck completed, returning 0
[Kasterisk1*CLI>
-- Executing NoOp("SIP/5060-b72a8b70", "No recording needed") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/5060-b72a8b70", "0?skipoutcid") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b72a8b70", "DIAL_TRUNK_OPTIONS=") in new stack
[Kasterisk1*CLI>
-- Executing Macro("SIP/5060-b72a8b70", "outbound-callerid|10") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/5060-b72a8b70", "1?start") in new stack
[Kasterisk1*CLI>
-- Goto (macro-outbound-callerid,s,3)
-- Executing NoOp("SIP/5060-b72a8b70", "REALCALLERIDNUM is 9006") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/5060-b72a8b70", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b72a8b70", "USEROUTCID=") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b72a8b70", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/5060-b72a8b70", "TRUNKOUTCID=") in new stack
-- Executing GotoIf("SIP/5060-b72a8b70", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing GotoIf("SIP/5060-b72a8b70", "1?usercid") in new stack
-- Goto (macro-outbound-callerid,s,18)
-- Executing GotoIf("SIP/5060-b72a8b70", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing NoOp("SIP/5060-b72a8b70", "CallerID set to "" <9006>") in new stack
-- Executing GotoIf("SIP/5060-b72a8b70", "0?nomax") in new stack
-- Executing GotoIf("SIP/5060-b72a8b70", "0?chanfull") in new stack
-- Executing DeadAGI("SIP/5060-b72a8b70", "fixlocalprefix") in new stack
[Kasterisk1*CLI>
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
[Kasterisk1*CLI>
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/5060-b72a8b70", "OUTNUM=9086455009") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b72a8b70", "custom=IAX2/GTT_No_VPN") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/5060-b72a8b70", "0?customtrunk") in new stack
-- Executing Set("SIP/5060-b72a8b70", "CDR(userfield)=204.14.9.50") in new stack
[Kasterisk1*CLI>
-- Executing Dial("SIP/5060-b72a8b70", "IAX2/GTT_No_VPN/9086455009|300|") in new stack
[Kasterisk1*CLI>
-- Called GTT_No_VPN/9086455009
[Kasterisk1*CLI>
-- Hungup 'IAX2/GTT_No_VPN-16416'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/5060-b720cfe0' in macro 'dialout-trunk'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/5060-b720cfe0'
-- Executing Macro("SIP/5060-b720cfe0", "hangupcall") in new stack
-- Executing ResetCDR("SIP/5060-b720cfe0", "w") in new stack
[Kasterisk1*CLI>
-- Executing NoCDR("SIP/5060-b720cfe0", "") in new stack
-- Executing GotoIf("SIP/5060-b720cfe0", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing GotoIf("SIP/5060-b720cfe0", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing Wait("SIP/5060-b720cfe0", "5") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/5060-b720cfe0' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/5060-b720cfe0'
[Kasterisk1*CLI>
-- Call accepted by 190.80.53.168 (format g723)
-- Format for call is g723
[Kasterisk1*CLI>
-- Hungup 'IAX2/GTT_No_VPN-16422'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/64.15.135.252-b72e2d80' in macro 'dialout-trunk'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/64.15.135.252-b72e2d80'
[Kasterisk1*CLI>
-- Executing Macro("SIP/64.15.135.252-b72e2d80", "hangupcall") in new stack
[Kasterisk1*CLI>
-- Executing ResetCDR("SIP/64.15.135.252-b72e2d80", "w") in new stack
[Kasterisk1*CLI>
-- Executing NoCDR("SIP/64.15.135.252-b72e2d80", "") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/64.15.135.252-b72e2d80", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing GotoIf("SIP/64.15.135.252-b72e2d80", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
[Kasterisk1*CLI>
-- Executing Wait("SIP/64.15.135.252-b72e2d80", "5") in new stack
[Kasterisk1*CLI>
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/64.15.135.252-b72e2d80' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/64.15.135.252-b72e2d80'
[Kasterisk1*CLI>
-- Hungup 'IAX2/Vydall-16396'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/64.15.135.252-b72eae48' in macro 'dialout-trunk'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/64.15.135.252-b72eae48'
-- Executing Macro("SIP/64.15.135.252-b72eae48", "hangupcall") in new stack
-- Executing ResetCDR("SIP/64.15.135.252-b72eae48", "w") in new stack
[Kasterisk1*CLI>
-- Executing NoCDR("SIP/64.15.135.252-b72eae48", "") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/64.15.135.252-b72eae48", "1?skiprg") in new stack
[Kasterisk1*CLI>
-- Goto (macro-hangupcall,s,6)
-- Executing GotoIf("SIP/64.15.135.252-b72eae48", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing Wait("SIP/64.15.135.252-b72eae48", "5") in new stack
[Kasterisk1*CLI>
-- Executing Macro("SIP/5060-b70a25b8", "dialout-trunk|10|9082183537||") in new stack
-- Executing Set("SIP/5060-b70a25b8", "CDR(userfield)=70.42.39.67") in new stack
-- Executing Set("SIP/5060-b70a25b8", "DIAL_TRUNK=10") in new stack
-- Executing Set("SIP/5060-b70a25b8", "_NODEST=") in new stack
-- Executing Set("SIP/5060-b70a25b8", "DIAL_NUMBER=9082183537") in new stack
-- Executing Set("SIP/5060-b70a25b8", "ROUTE_PASSWD=") in new stack
-- Executing Set("SIP/5060-b70a25b8", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing GotoIf("SIP/5060-b70a25b8", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing Set("SIP/5060-b70a25b8", "GROUP()=OUT_10") in new stack
-- Executing Macro("SIP/5060-b70a25b8", "user-callerid|SKIPTTL") in new stack
-- Executing NoOp("SIP/5060-b70a25b8", "user-callerid: 9010") in new stack
-- Executing GotoIf("SIP/5060-b70a25b8", "0?report") in new stack
-- Executing GotoIf("SIP/5060-b70a25b8", "0?start") in new stack
-- Executing Set("SIP/5060-b70a25b8", "REALCALLERIDNUM=9010") in new stack
-- Executing NoOp("SIP/5060-b70a25b8", "REALCALLERIDNUM is 9010") in new stack
-- Executing Set("SIP/5060-b70a25b8", "AMPUSER=") in new stack
-- Executing Set("SIP/5060-b70a25b8", "AMPUSERCIDNAME=") in new stack
-- Executing GotoIf("SIP/5060-b70a25b8", "1?report") in new stack
-- Goto (macro-user-callerid,s,11)
-- Executing NoOp("SIP/5060-b70a25b8", "TTL: ARG1: SKIPTTL") in new stack
-- Executing GotoIf("SIP/5060-b70a25b8", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/5060-b70a25b8", "Using CallerID "" <9010>") in new stack
-- Executing Macro("SIP/5060-b70a25b8", "record-enable|9010|OUT") in new stack
-- Executing GotoIf("SIP/5060-b70a25b8", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("SIP/5060-b70a25b8", "recordingcheck|20071109-015912|1194573552.135427") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
[Kasterisk1*CLI>
recordingcheck|20071109-015912|1194573552.135427: No AMPUSER db entry for 9010. Not recording
[Kasterisk1*CLI>
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/5060-b70a25b8", "No recording needed") in new stack
-- Executing GotoIf("SIP/5060-b70a25b8", "0?skipoutcid") in new stack
-- Executing Set("SIP/5060-b70a25b8", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing Macro("SIP/5060-b70a25b8", "outbound-callerid|10") in new stack
-- Executing GotoIf("SIP/5060-b70a25b8", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing NoOp("SIP/5060-b70a25b8", "REALCALLERIDNUM is 9010") in new stack
-- Executing GotoIf("SIP/5060-b70a25b8", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing Set("SIP/5060-b70a25b8", "USEROUTCID=") in new stack
-- Executing Set("SIP/5060-b70a25b8", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/5060-b70a25b8", "TRUNKOUTCID=") in new stack
-- Executing GotoIf("SIP/5060-b70a25b8", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing GotoIf("SIP/5060-b70a25b8", "1?usercid") in new stack
-- Goto (macro-outbound-callerid,s,18)
-- Executing GotoIf("SIP/5060-b70a25b8", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing NoOp("SIP/5060-b70a25b8", "CallerID set to "" <9010>") in new stack
-- Executing GotoIf("SIP/5060-b70a25b8", "0?nomax") in new stack
-- Executing GotoIf("SIP/5060-b70a25b8", "0?chanfull") in new stack
-- Executing DeadAGI("SIP/5060-b70a25b8", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
[Kasterisk1*CLI>
-- IAX2/GTT_No_VPN-16418 is ringing
[Kasterisk1*CLI>
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/5060-b70a25b8", "OUTNUM=9082183537") in new stack
-- Executing Set("SIP/5060-b70a25b8", "custom=IAX2/GTT_No_VPN") in new stack
-- Executing GotoIf("SIP/5060-b70a25b8", "0?customtrunk") in new stack
-- Executing Set("SIP/5060-b70a25b8", "CDR(userfield)=70.42.39.67") in new stack
-- Executing Dial("SIP/5060-b70a25b8", "IAX2/GTT_No_VPN/9082183537|300|") in new stack
-- Called GTT_No_VPN/9082183537
[Kasterisk1*CLI>
-- Call accepted by 190.80.53.168 (format g723)
-- Format for call is g723
[Kasterisk1*CLI>
-- IAX2/GTT_No_VPN-16419 is ringing
[Kasterisk1*CLI>
-- Executing Hangup("SIP/64.15.135.252-b72eae48", "") in new stack
== Spawn extension (macro-hangupcall, s, 10) exited non-zero on 'SIP/64.15.135.252-b72eae48' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 10) exited non-zero on 'SIP/64.15.135.252-b72eae48'
[Kasterisk1*CLI>
-- Executing Macro("SIP/5060-b729c010", "dialout-trunk|10|9086417180||") in new stack
-- Executing Set("SIP/5060-b729c010", "CDR(userfield)=204.14.9.50") in new stack
-- Executing Set("SIP/5060-b729c010", "DIAL_TRUNK=10") in new stack
-- Executing Set("SIP/5060-b729c010", "_NODEST=") in new stack
-- Executing Set("SIP/5060-b729c010", "DIAL_NUMBER=9086417180") in new stack
-- Executing Set("SIP/5060-b729c010", "ROUTE_PASSWD=") in new stack
-- Executing Set("SIP/5060-b729c010", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing GotoIf("SIP/5060-b729c010", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing Set("SIP/5060-b729c010", "GROUP()=OUT_10") in new stack
-- Executing Macro("SIP/5060-b729c010", "user-callerid|SKIPTTL") in new stack
-- Executing NoOp("SIP/5060-b729c010", "user-callerid: ") in new stack
-- Executing GotoIf("SIP/5060-b729c010", "0?report") in new stack
-- Executing GotoIf("SIP/5060-b729c010", "0?start") in new stack
-- Executing Set("SIP/5060-b729c010", "REALCALLERIDNUM=") in new stack
-- Executing NoOp("SIP/5060-b729c010", "REALCALLERIDNUM is ") in new stack
-- Executing Set("SIP/5060-b729c010", "AMPUSER=") in new stack
-- Executing Set("SIP/5060-b729c010", "AMPUSERCIDNAME=") in new stack
-- Executing GotoIf("SIP/5060-b729c010", "1?report") in new stack
-- Goto (macro-user-callerid,s,11)
-- Executing NoOp("SIP/5060-b729c010", "TTL: ARG1: SKIPTTL") in new stack
-- Executing GotoIf("SIP/5060-b729c010", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/5060-b729c010", "Using CallerID "" <>") in new stack
-- Executing Macro("SIP/5060-b729c010", "record-enable||OUT") in new stack
-- Executing GotoIf("SIP/5060-b729c010", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("SIP/5060-b729c010", "recordingcheck|20071109-015917|1194573557.135429") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
[Kasterisk1*CLI>
recordingcheck|20071109-015917|1194573557.135429: No AMPUSER db entry for . Not recording
[Kasterisk1*CLI>
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/5060-b729c010", "No recording needed") in new stack
-- Executing GotoIf("SIP/5060-b729c010", "0?skipoutcid") in new stack
-- Executing Set("SIP/5060-b729c010", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing Macro("SIP/5060-b729c010", "outbound-callerid|10") in new stack
-- Executing GotoIf("SIP/5060-b729c010", "0?start") in new stack
-- Executing Set("SIP/5060-b729c010", "REALCALLERIDNUM=") in new stack
-- Executing NoOp("SIP/5060-b729c010", "REALCALLERIDNUM is ") in new stack
-- Executing GotoIf("SIP/5060-b729c010", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing Set("SIP/5060-b729c010", "USEROUTCID=") in new stack
-- Executing Set("SIP/5060-b729c010", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/5060-b729c010", "TRUNKOUTCID=") in new stack
-- Executing GotoIf("SIP/5060-b729c010", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing GotoIf("SIP/5060-b729c010", "1?usercid") in new stack
-- Goto (macro-outbound-callerid,s,18)
-- Executing GotoIf("SIP/5060-b729c010", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing NoOp("SIP/5060-b729c010", "CallerID set to "" <>") in new stack
-- Executing GotoIf("SIP/5060-b729c010", "0?nomax") in new stack
-- Executing GotoIf("SIP/5060-b729c010", "0?chanfull") in new stack
-- Executing DeadAGI("SIP/5060-b729c010", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
[Kasterisk1*CLI>
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/5060-b729c010", "OUTNUM=9086417180") in new stack
-- Executing Set("SIP/5060-b729c010", "custom=IAX2/GTT_No_VPN") in new stack
-- Executing GotoIf("SIP/5060-b729c010", "0?customtrunk") in new stack
-- Executing Set("SIP/5060-b729c010", "CDR(userfield)=204.14.9.50") in new stack
-- Executing Dial("SIP/5060-b729c010", "IAX2/GTT_No_VPN/9086417180|300|") in new stack
-- Called GTT_No_VPN/9086417180
[Kasterisk1*CLI>
-- Hungup 'IAX2/Vydall-16398'
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Goto("SIP/5060-b7219088", "s-CHANUNAVAIL|1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing NoOp("SIP/5060-b7219088", "Dial failed due to CHANUNAVAIL - failing through to other trunks") in new stack
-- Executing Macro("SIP/5060-b7219088", "dialout-trunk|9|9082655827||") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b7219088", "CDR(userfield)=204.14.9.50") in new stack
-- Executing Set("SIP/5060-b7219088", "DIAL_TRUNK=9") in new stack
-- Executing Set("SIP/5060-b7219088", "_NODEST=") in new stack
-- Executing Set("SIP/5060-b7219088", "DIAL_NUMBER=9082655827") in new stack
-- Executing Set("SIP/5060-b7219088", "ROUTE_PASSWD=") in new stack
-- Executing Set("SIP/5060-b7219088", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing Set("SIP/5060-b7219088", "GROUP()=OUT_9") in new stack
-- Executing Macro("SIP/5060-b7219088", "user-callerid|SKIPTTL") in new stack
-- Executing NoOp("SIP/5060-b7219088", "user-callerid: 9006") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "0?report") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "1?start") in new stack
-- Goto (macro-user-callerid,s,5)
-- Executing NoOp("SIP/5060-b7219088", "REALCALLERIDNUM is 9006") in new stack
-- Executing Set("SIP/5060-b7219088", "AMPUSER=") in new stack
-- Executing Set("SIP/5060-b7219088", "AMPUSERCIDNAME=") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "1?report") in new stack
-- Goto (macro-user-callerid,s,11)
-- Executing NoOp("SIP/5060-b7219088", "TTL: ARG1: SKIPTTL") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/5060-b7219088", "Using CallerID "" <9006>") in new stack
-- Executing Macro("SIP/5060-b7219088", "record-enable|9006|OUT") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("SIP/5060-b7219088", "recordingcheck|20071109-015917|1194573524.135408") in new stack
[Kasterisk1*CLI>
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
[Kasterisk1*CLI>
-- Call accepted by 190.80.53.168 (format g723)
-- Format for call is g723
[Kasterisk1*CLI>
recordingcheck|20071109-015917|1194573524.135408: No AMPUSER db entry for 9006. Not recording
[Kasterisk1*CLI>
-- AGI Script recordingcheck completed, returning 0
[Kasterisk1*CLI>
-- Executing NoOp("SIP/5060-b7219088", "No recording needed") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/5060-b7219088", "0?skipoutcid") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b7219088", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing Macro("SIP/5060-b7219088", "outbound-callerid|9") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/5060-b7219088", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing NoOp("SIP/5060-b7219088", "REALCALLERIDNUM is 9006") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing Set("SIP/5060-b7219088", "USEROUTCID=") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b7219088", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/5060-b7219088", "TRUNKOUTCID=") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing GotoIf("SIP/5060-b7219088", "1?usercid") in new stack
-- Goto (macro-outbound-callerid,s,18)
-- Executing GotoIf("SIP/5060-b7219088", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing NoOp("SIP/5060-b7219088", "CallerID set to "" <9006>") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "0?nomax") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "0?chanfull") in new stack
[Kasterisk1*CLI>
-- Executing DeadAGI("SIP/5060-b7219088", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
[Kasterisk1*CLI>
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/5060-b7219088", "OUTNUM=9082655827") in new stack
-- Executing Set("SIP/5060-b7219088", "custom=SIP/Addpac") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "0?customtrunk") in new stack
-- Executing Set("SIP/5060-b7219088", "CDR(userfield)=204.14.9.50") in new stack
-- Executing Dial("SIP/5060-b7219088", "SIP/Addpac/9082655827|300|") in new stack
-- Called Addpac/9082655827
[Kasterisk1*CLI>
-- Hungup 'IAX2/GTT_No_VPN-16407'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/64.15.135.252-b729f6f8' in macro 'dialout-trunk'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/64.15.135.252-b729f6f8'
-- Executing Macro("SIP/64.15.135.252-b729f6f8", "hangupcall") in new stack
-- Executing ResetCDR("SIP/64.15.135.252-b729f6f8", "w") in new stack
[Kasterisk1*CLI>
-- Executing NoCDR("SIP/64.15.135.252-b729f6f8", "") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b729f6f8", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing GotoIf("SIP/64.15.135.252-b729f6f8", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing Wait("SIP/64.15.135.252-b729f6f8", "5") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/64.15.135.252-b729f6f8' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/64.15.135.252-b729f6f8'
[Kasterisk1*CLI>
-- SIP/Addpac-b70bb9a0 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Goto("SIP/5060-b7219088", "s-CONGESTION|1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing NoOp("SIP/5060-b7219088", "Dial failed due to CONGESTION - failing through to other trunks") in new stack
[Kasterisk1*CLI>
-- Executing Macro("SIP/5060-b7219088", "outisbusy|") in new stack
[Kasterisk1*CLI>
-- Executing Playback("SIP/5060-b7219088", "all-circuits-busy-now|noanswer") in new stack
[Kasterisk1*CLI>
-- Playing 'all-circuits-busy-now' (language 'en')
[Kasterisk1*CLI>
-- Hungup 'IAX2/GTT_No_VPN-16392'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/66.246.197.20-b7083060' in macro 'dialout-trunk'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/66.246.197.20-b7083060'
-- Executing Macro("SIP/66.246.197.20-b7083060", "hangupcall") in new stack
-- Executing ResetCDR("SIP/66.246.197.20-b7083060", "w") in new stack
[Kasterisk1*CLI>
-- Executing NoCDR("SIP/66.246.197.20-b7083060", "") in new stack
-- Executing GotoIf("SIP/66.246.197.20-b7083060", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing GotoIf("SIP/66.246.197.20-b7083060", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing Wait("SIP/66.246.197.20-b7083060", "5") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/66.246.197.20-b7083060' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/66.246.197.20-b7083060'
== End MixMonitor Recording SIP/66.246.197.20-b7083060
[Kasterisk1*CLI>
Destroying call '2C1CA2B1-324711D6-98C6F52C-1F15DBCD@64.192.196.12'
[Kasterisk1*CLI>
== Spawn extension (macro-outisbusy, s, 1) exited non-zero on 'SIP/5060-b7219088' in macro 'outisbusy'
== Spawn extension (macro-outisbusy, s, 1) exited non-zero on 'SIP/5060-b7219088'
[Kasterisk1*CLI>
12 headers, 0 lines
Reliably Transmitting (NAT) to 64.192.196.12:5060:
OPTIONS sip:64.192.196.12 SIP/2.0
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK59df2c92;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as1a98338c
To: <sip:64.192.196.12>
Contact: <sip:Unknown@208.75.208.90>
Call-ID: 4d19c7a55b1a515952caa2522b4f771c@208.75.208.90
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 09 Nov 2007 01:59:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
[Kasterisk1*CLI>
<-- SIP read from 64.192.196.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK59df2c92;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as1a98338c
To: <sip:64.192.196.12>;tag=2A035F97-1178
Date: Sat, 09 Mar 2002 03:47:44 GMT
Call-ID: 4d19c7a55b1a515952caa2522b4f771c@208.75.208.90
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,resource-priority,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 451
v=0
o=CiscoSystemsSIP-GW-UserAgent 6473 6673 IN IP4 64.192.196.12
s=SIP Call
c=IN IP4 64.192.196.12
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 15 3
c=IN IP4 64.192.196.12
m=image 0 udptl t38
c=IN IP4 64.192.196.12
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
--- (14 headers 18 lines) ---
[Kasterisk1*CLI>
Destroying call '4d19c7a55b1a515952caa2522b4f771c@208.75.208.90'
[Kasterisk1*CLI>
12 headers, 0 lines
Reliably Transmitting (NAT) to 64.192.196.12:5060:
OPTIONS sip:64.192.196.12 SIP/2.0
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK22f376ca;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as56bc2d1e
To: <sip:64.192.196.12>
Contact: <sip:Unknown@208.75.208.90>
Call-ID: 4f38ba7c7ce36eb1461ec3456fc07f66@208.75.208.90
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 09 Nov 2007 01:59:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
[Kasterisk1*CLI>
<-- SIP read from 64.192.196.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK22f376ca;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as56bc2d1e
To: <sip:64.192.196.12>;tag=2A035FCF-2BC
Date: Sat, 09 Mar 2002 03:47:44 GMT
Call-ID: 4f38ba7c7ce36eb1461ec3456fc07f66@208.75.208.90
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,resource-priority,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 451
v=0
o=CiscoSystemsSIP-GW-UserAgent 5172 3149 IN IP4 64.192.196.12
s=SIP Call
c=IN IP4 64.192.196.12
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 15 3
c=IN IP4 64.192.196.12
m=image 0 udptl t38
c=IN IP4 64.192.196.12
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
[Kasterisk1*CLI>
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
--- (14 headers 18 lines) ---
[Kasterisk1*CLI>
Destroying call '4f38ba7c7ce36eb1461ec3456fc07f66@208.75.208.90'
[Kasterisk1*CLI>
-- Hungup 'IAX2/GTT_No_VPN-16414'
[Kasterisk1*CLI>
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/64.15.135.252-b722ed50' in macro 'dialout-trunk'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/64.15.135.252-b722ed50'
-- Executing Macro("SIP/64.15.135.252-b722ed50", "hangupcall") in new stack
[Kasterisk1*CLI>
-- Executing ResetCDR("SIP/64.15.135.252-b722ed50", "w") in new stack
[Kasterisk1*CLI>
-- Executing NoCDR("SIP/64.15.135.252-b722ed50", "") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b722ed50", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing GotoIf("SIP/64.15.135.252-b722ed50", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing Wait("SIP/64.15.135.252-b722ed50", "5") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/64.15.135.252-b722ed50' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/64.15.135.252-b722ed50'
[Kasterisk1*CLI>
-- Executing Macro("SIP/5060-b72e2d80", "dialout-trunk|16|7619048||") in new stack
-- Executing Set("SIP/5060-b72e2d80", "CDR(userfield)=66.241.7.200") in new stack
-- Executing Set("SIP/5060-b72e2d80", "DIAL_TRUNK=16") in new stack
-- Executing Set("SIP/5060-b72e2d80", "_NODEST=") in new stack
-- Executing Set("SIP/5060-b72e2d80", "DIAL_NUMBER=7619048") in new stack
-- Executing Set("SIP/5060-b72e2d80", "ROUTE_PASSWD=") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b72e2d80", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing GotoIf("SIP/5060-b72e2d80", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,9)
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b72e2d80", "GROUP()=OUT_16") in new stack
-- Executing Macro("SIP/5060-b72e2d80", "user-callerid|SKIPTTL") in new stack
-- Executing NoOp("SIP/5060-b72e2d80", "user-callerid: Vincom ") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/5060-b72e2d80", "0?report") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/5060-b72e2d80", "0?start") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b72e2d80", "REALCALLERIDNUM=") in new stack
[Kasterisk1*CLI>
-- Executing NoOp("SIP/5060-b72e2d80", "REALCALLERIDNUM is ") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b72e2d80", "AMPUSER=") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b72e2d80", "AMPUSERCIDNAME=") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/5060-b72e2d80", "1?report") in new stack
[Kasterisk1*CLI>
-- Goto (macro-user-callerid,s,11)
-- Executing NoOp("SIP/5060-b72e2d80", "TTL: ARG1: SKIPTTL") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/5060-b72e2d80", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/5060-b72e2d80", "Using CallerID "Vincom" <>") in new stack
-- Executing Macro("SIP/5060-b72e2d80", "record-enable||OUT") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/5060-b72e2d80", "0 > 0?2:4") in new stack
[Kasterisk1*CLI>
-- Goto (macro-record-enable,s,4)
[Kasterisk1*CLI>
-- Executing DeadAGI("SIP/5060-b72e2d80", "recordingcheck|20071109-019081|1194573561.135432") in new stack
[Kasterisk1*CLI>
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
[Kasterisk1*CLI>
recordingcheck|20071109-019081|1194573561.135432: No AMPUSER db entry for . Not recording
[Kasterisk1*CLI>
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/5060-b72e2d80", "No recording needed") in new stack
-- Executing GotoIf("SIP/5060-b72e2d80", "0?skipoutcid") in new stack
-- Executing Set("SIP/5060-b72e2d80", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing Macro("SIP/5060-b72e2d80", "outbound-callerid|16") in new stack
-- Executing GotoIf("SIP/5060-b72e2d80", "0?start") in new stack
-- Executing Set("SIP/5060-b72e2d80", "REALCALLERIDNUM=") in new stack
-- Executing NoOp("SIP/5060-b72e2d80", "REALCALLERIDNUM is ") in new stack
-- Executing GotoIf("SIP/5060-b72e2d80", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing Set("SIP/5060-b72e2d80", "USEROUTCID=") in new stack
-- Executing Set("SIP/5060-b72e2d80", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/5060-b72e2d80", "TRUNKOUTCID=") in new stack
-- Executing GotoIf("SIP/5060-b72e2d80", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing GotoIf("SIP/5060-b72e2d80", "1?usercid") in new stack
-- Goto (macro-outbound-callerid,s,18)
-- Executing GotoIf("SIP/5060-b72e2d80", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing NoOp("SIP/5060-b72e2d80", "CallerID set to "Vincom" <>") in new stack
-- Executing GotoIf("SIP/5060-b72e2d80", "0?nomax") in new stack
-- Executing GotoIf("SIP/5060-b72e2d80", "0?chanfull") in new stack
-- Executing DeadAGI("SIP/5060-b72e2d80", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
[Kasterisk1*CLI>
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/5060-b72e2d80", "OUTNUM=7619048") in new stack
-- Executing Set("SIP/5060-b72e2d80", "custom=IAX2/Adesh_Trinidad") in new stack
-- Executing GotoIf("SIP/5060-b72e2d80", "0?customtrunk") in new stack
-- Executing Set("SIP/5060-b72e2d80", "CDR(userfield)=66.241.7.200") in new stack
-- Executing Dial("SIP/5060-b72e2d80", "IAX2/Adesh_Trinidad/7619048|300|") in new stack
-- Called Adesh_Trinidad/7619048
[Kasterisk1*CLI>
-- Call accepted by 201.238.76.182 (format g723)
-- Format for call is g723
[Kasterisk1*CLI>
-- IAX2/Adesh_Trinidad-16423 is ringing
[Kasterisk1*CLI>
-- IAX2/GTT_No_VPN-16427 is ringing
[Kasterisk1*CLI>
-- IAX2/GTT_No_VPN-16415 is ringing
[Kasterisk1*CLI>
-- Executing Macro("SIP/5060-b7219088", "dialout-trunk|10|9082642679||") in new stack
-- Executing Set("SIP/5060-b7219088", "CDR(userfield)=204.14.9.50") in new stack
-- Executing Set("SIP/5060-b7219088", "DIAL_TRUNK=10") in new stack
-- Executing Set("SIP/5060-b7219088", "_NODEST=") in new stack
-- Executing Set("SIP/5060-b7219088", "DIAL_NUMBER=9082642679") in new stack
-- Executing Set("SIP/5060-b7219088", "ROUTE_PASSWD=") in new stack
-- Executing Set("SIP/5060-b7219088", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing Set("SIP/5060-b7219088", "GROUP()=OUT_10") in new stack
-- Executing Macro("SIP/5060-b7219088", "user-callerid|SKIPTTL") in new stack
-- Executing NoOp("SIP/5060-b7219088", "user-callerid: NextC 9006") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "0?report") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "0?start") in new stack
-- Executing Set("SIP/5060-b7219088", "REALCALLERIDNUM=9006") in new stack
-- Executing NoOp("SIP/5060-b7219088", "REALCALLERIDNUM is 9006") in new stack
-- Executing Set("SIP/5060-b7219088", "AMPUSER=") in new stack
-- Executing Set("SIP/5060-b7219088", "AMPUSERCIDNAME=") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "1?report") in new stack
-- Goto (macro-user-callerid,s,11)
-- Executing NoOp("SIP/5060-b7219088", "TTL: ARG1: SKIPTTL") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/5060-b7219088", "Using CallerID "NextC" <9006>") in new stack
-- Executing Macro("SIP/5060-b7219088", "record-enable|9006|OUT") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("SIP/5060-b7219088", "recordingcheck|20071109-019085|1194573565.135434") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
[Kasterisk1*CLI>
recordingcheck|20071109-019085|1194573565.135434: No AMPUSER db entry for 9006. Not recording
[Kasterisk1*CLI>
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/5060-b7219088", "No recording needed") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "0?skipoutcid") in new stack
-- Executing Set("SIP/5060-b7219088", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing Macro("SIP/5060-b7219088", "outbound-callerid|10") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing NoOp("SIP/5060-b7219088", "REALCALLERIDNUM is 9006") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing Set("SIP/5060-b7219088", "USEROUTCID=") in new stack
-- Executing Set("SIP/5060-b7219088", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/5060-b7219088", "TRUNKOUTCID=") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing GotoIf("SIP/5060-b7219088", "1?usercid") in new stack
-- Goto (macro-outbound-callerid,s,18)
-- Executing GotoIf("SIP/5060-b7219088", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing NoOp("SIP/5060-b7219088", "CallerID set to "NextC" <9006>") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "0?nomax") in new stack
-- Executing GotoIf("SIP/5060-b7219088", "0?chanfull") in new stack
-- Executing DeadAGI("SIP/5060-b7219088", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
[Kasterisk1*CLI>
-- AGI Script fixlocalprefix completed, returning 0
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b7219088", "OUTNUM=9082642679") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b7219088", "custom=IAX2/GTT_No_VPN") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/5060-b7219088", "0?customtrunk") in new stack
-- Executing Set("SIP/5060-b7219088", "CDR(userfield)=204.14.9.50") in new stack
[Kasterisk1*CLI>
-- Executing Dial("SIP/5060-b7219088", "IAX2/GTT_No_VPN/9082642679|300|") in new stack
[Kasterisk1*CLI>
-- Called GTT_No_VPN/9082642679
[Kasterisk1*CLI>
-- IAX2/GTT_No_VPN-16418 stopped sounds
[Kasterisk1*CLI>
-- IAX2/GTT_No_VPN-16418 answered SIP/5060-b70a51d8
[Kasterisk1*CLI>
-- Call accepted by 190.80.53.168 (format g723)
-- Format for call is g723
[Kasterisk1*CLI>
-- Executing Macro("SIP/66.246.197.20-b7083060", "dialout-trunk|9|9082271111||") in new stack
-- Executing Set("SIP/66.246.197.20-b7083060", "CDR(userfield)=66.246.197.20") in new stack
-- Executing Set("SIP/66.246.197.20-b7083060", "DIAL_TRUNK=9") in new stack
-- Executing Set("SIP/66.246.197.20-b7083060", "_NODEST=") in new stack
-- Executing Set("SIP/66.246.197.20-b7083060", "DIAL_NUMBER=9082271111") in new stack
-- Executing Set("SIP/66.246.197.20-b7083060", "ROUTE_PASSWD=") in new stack
-- Executing Set("SIP/66.246.197.20-b7083060", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing GotoIf("SIP/66.246.197.20-b7083060", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing Set("SIP/66.246.197.20-b7083060", "GROUP()=OUT_9") in new stack
-- Executing Macro("SIP/66.246.197.20-b7083060", "user-callerid|SKIPTTL") in new stack
-- Executing NoOp("SIP/66.246.197.20-b7083060", "user-callerid: 9002") in new stack
-- Executing GotoIf("SIP/66.246.197.20-b7083060", "0?report") in new stack
-- Executing GotoIf("SIP/66.246.197.20-b7083060", "0?start") in new stack
-- Executing Set("SIP/66.246.197.20-b7083060", "REALCALLERIDNUM=9002") in new stack
-- Executing NoOp("SIP/66.246.197.20-b7083060", "REALCALLERIDNUM is 9002") in new stack
-- Executing Set("SIP/66.246.197.20-b7083060", "AMPUSER=9002") in new stack
-- Executing Set("SIP/66.246.197.20-b7083060", "AMPUSERCIDNAME=mark") in new stack
-- Executing GotoIf("SIP/66.246.197.20-b7083060", "0?report") in new stack
-- Executing Set("SIP/66.246.197.20-b7083060", "CALLERID(all)=mark <9002>") in new stack
-- Executing Set("SIP/66.246.197.20-b7083060", "REALCALLERIDNUM=9002") in new stack
-- Executing NoOp("SIP/66.246.197.20-b7083060", "TTL: ARG1: SKIPTTL") in new stack
-- Executing GotoIf("SIP/66.246.197.20-b7083060", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/66.246.197.20-b7083060", "Using CallerID "mark" <9002>") in new stack
-- Executing Macro("SIP/66.246.197.20-b7083060", "record-enable|9002|OUT") in new stack
-- Executing GotoIf("SIP/66.246.197.20-b7083060", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("SIP/66.246.197.20-b7083060", "recordingcheck|20071109-019089|1194573569.135436") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
[Kasterisk1*CLI>
recordingcheck|20071109-019089|1194573569.135436: Outbound recording enabled.
[Kasterisk1*CLI>
recordingcheck|20071109-019089|1194573569.135436: CALLFILENAME=OUT9002-20071109-019089-1194573569.135436
[Kasterisk1*CLI>
-- AGI Script recordingcheck completed, returning 0
-- Executing MixMonitor("SIP/66.246.197.20-b7083060", "OUT9002-20071109-019089-1194573569.135436.wav") in new stack
-- Executing GotoIf("SIP/66.246.197.20-b7083060", "0?skipoutcid") in new stack
-- Executing Set("SIP/66.246.197.20-b7083060", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing Macro("SIP/66.246.197.20-b7083060", "outbound-callerid|9") in new stack
-- Executing GotoIf("SIP/66.246.197.20-b7083060", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
[Kasterisk1*CLI>
-- Executing NoOp("SIP/66.246.197.20-b7083060", "REALCALLERIDNUM is 9002") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/66.246.197.20-b7083060", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing Set("SIP/66.246.197.20-b7083060", "USEROUTCID=") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/66.246.197.20-b7083060", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/66.246.197.20-b7083060", "TRUNKOUTCID=") in new stack
-- Executing GotoIf("SIP/66.246.197.20-b7083060", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing GotoIf("SIP/66.246.197.20-b7083060", "1?usercid") in new stack
-- Goto (macro-outbound-callerid,s,18)
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/66.246.197.20-b7083060", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing NoOp("SIP/66.246.197.20-b7083060", "CallerID set to "mark" <9002>") in new stack
-- Executing GotoIf("SIP/66.246.197.20-b7083060", "0?nomax") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/66.246.197.20-b7083060", "0?chanfull") in new stack
-- Executing DeadAGI("SIP/66.246.197.20-b7083060", "fixlocalprefix") in new stack
[Kasterisk1*CLI>
== Begin MixMonitor Recording SIP/66.246.197.20-b7083060
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
[Kasterisk1*CLI>
-- Executing Macro("SIP/66.246.197.20-b70257d8", "dialout-trunk|9|9082206980||") in new stack
-- Executing Set("SIP/66.246.197.20-b70257d8", "CDR(userfield)=66.246.197.20") in new stack
-- Executing Set("SIP/66.246.197.20-b70257d8", "DIAL_TRUNK=9") in new stack
-- Executing Set("SIP/66.246.197.20-b70257d8", "_NODEST=") in new stack
-- Executing Set("SIP/66.246.197.20-b70257d8", "DIAL_NUMBER=9082206980") in new stack
-- Executing Set("SIP/66.246.197.20-b70257d8", "ROUTE_PASSWD=") in new stack
-- Executing Set("SIP/66.246.197.20-b70257d8", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing GotoIf("SIP/66.246.197.20-b70257d8", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing Set("SIP/66.246.197.20-b70257d8", "GROUP()=OUT_9") in new stack
-- Executing Macro("SIP/66.246.197.20-b70257d8", "user-callerid|SKIPTTL") in new stack
-- Executing NoOp("SIP/66.246.197.20-b70257d8", "user-callerid: 9002") in new stack
-- Executing GotoIf("SIP/66.246.197.20-b70257d8", "0?report") in new stack
-- Executing GotoIf("SIP/66.246.197.20-b70257d8", "0?start") in new stack
-- Executing Set("SIP/66.246.197.20-b70257d8", "REALCALLERIDNUM=9002") in new stack
-- Executing NoOp("SIP/66.246.197.20-b70257d8", "REALCALLERIDNUM is 9002") in new stack
-- Executing Set("SIP/66.246.197.20-b70257d8", "AMPUSER=9002") in new stack
-- Executing Set("SIP/66.246.197.20-b70257d8", "AMPUSERCIDNAME=mark") in new stack
-- Executing GotoIf("SIP/66.246.197.20-b70257d8", "0?report") in new stack
-- Executing Set("SIP/66.246.197.20-b70257d8", "CALLERID(all)=mark <9002>") in new stack
-- Executing Set("SIP/66.246.197.20-b70257d8", "REALCALLERIDNUM=9002") in new stack
-- Executing NoOp("SIP/66.246.197.20-b70257d8", "TTL: ARG1: SKIPTTL") in new stack
-- Executing GotoIf("SIP/66.246.197.20-b70257d8", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/66.246.197.20-b70257d8", "Using CallerID "mark" <9002>") in new stack
-- Executing Macro("SIP/66.246.197.20-b70257d8", "record-enable|9002|OUT") in new stack
-- Executing GotoIf("SIP/66.246.197.20-b70257d8", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("SIP/66.246.197.20-b70257d8", "recordingcheck|20071109-019089|1194573569.135437") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
[Kasterisk1*CLI>
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/66.246.197.20-b7083060", "OUTNUM=9082271111") in new stack
-- Executing Set("SIP/66.246.197.20-b7083060", "custom=SIP/Addpac") in new stack
-- Executing GotoIf("SIP/66.246.197.20-b7083060", "0?customtrunk") in new stack
-- Executing Set("SIP/66.246.197.20-b7083060", "CDR(userfield)=66.246.197.20") in new stack
-- Executing Dial("SIP/66.246.197.20-b7083060", "SIP/Addpac/9082271111|300|") in new stack
-- Called Addpac/9082271111
[Kasterisk1*CLI>
recordingcheck|20071109-019089|1194573569.135437: Outbound recording enabled.
[Kasterisk1*CLI>
recordingcheck|20071109-019089|1194573569.135437: CALLFILENAME=OUT9002-20071109-019089-1194573569.135437
[Kasterisk1*CLI>
-- AGI Script recordingcheck completed, returning 0
-- Executing MixMonitor("SIP/66.246.197.20-b70257d8", "OUT9002-20071109-019089-1194573569.135437.wav") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/66.246.197.20-b70257d8", "0?skipoutcid") in new stack
== Begin MixMonitor Recording SIP/66.246.197.20-b70257d8
[Kasterisk1*CLI>
-- Executing Set("SIP/66.246.197.20-b70257d8", "DIAL_TRUNK_OPTIONS=") in new stack
[Kasterisk1*CLI>
-- Executing Macro("SIP/66.246.197.20-b70257d8", "outbound-callerid|9") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/66.246.197.20-b70257d8", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
[Kasterisk1*CLI>
-- Executing NoOp("SIP/66.246.197.20-b70257d8", "REALCALLERIDNUM is 9002") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/66.246.197.20-b70257d8", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing Set("SIP/66.246.197.20-b70257d8", "USEROUTCID=") in new stack
-- Executing Set("SIP/66.246.197.20-b70257d8", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/66.246.197.20-b70257d8", "TRUNKOUTCID=") in new stack
-- Executing GotoIf("SIP/66.246.197.20-b70257d8", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing GotoIf("SIP/66.246.197.20-b70257d8", "1?usercid") in new stack
-- Goto (macro-outbound-callerid,s,18)
-- Executing GotoIf("SIP/66.246.197.20-b70257d8", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing NoOp("SIP/66.246.197.20-b70257d8", "CallerID set to "mark" <9002>") in new stack
-- Executing GotoIf("SIP/66.246.197.20-b70257d8", "0?nomax") in new stack
-- Executing GotoIf("SIP/66.246.197.20-b70257d8", "0?chanfull") in new stack
-- Executing DeadAGI("SIP/66.246.197.20-b70257d8", "fixlocalprefix") in new stack
[Kasterisk1*CLI>
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
[Kasterisk1*CLI>
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/66.246.197.20-b70257d8", "OUTNUM=9082206980") in new stack
-- Executing Set("SIP/66.246.197.20-b70257d8", "custom=SIP/Addpac") in new stack
-- Executing GotoIf("SIP/66.246.197.20-b70257d8", "0?customtrunk") in new stack
-- Executing Set("SIP/66.246.197.20-b70257d8", "CDR(userfield)=66.246.197.20") in new stack
-- Executing Dial("SIP/66.246.197.20-b70257d8", "SIP/Addpac/9082206980|300|") in new stack
-- Called Addpac/9082206980
[Kasterisk1*CLI>
-- SIP/Addpac-08856628 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Goto("SIP/66.246.197.20-b7083060", "s-CONGESTION|1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing NoOp("SIP/66.246.197.20-b7083060", "Dial failed due to CONGESTION - failing through to other trunks") in new stack
-- Executing Macro("SIP/66.246.197.20-b7083060", "outisbusy|") in new stack
-- Executing Playback("SIP/66.246.197.20-b7083060", "all-circuits-busy-now|noanswer") in new stack
-- Playing 'all-circuits-busy-now' (language 'en')
[Kasterisk1*CLI>
-- Executing Macro("SIP/5060-b709f998", "dialout-trunk|10|9086236547||") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b709f998", "CDR(userfield)=204.14.9.50") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b709f998", "DIAL_TRUNK=10") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b709f998", "_NODEST=") in new stack
-- Executing Set("SIP/5060-b709f998", "DIAL_NUMBER=9086236547") in new stack
-- Executing Set("SIP/5060-b709f998", "ROUTE_PASSWD=") in new stack
-- Executing Set("SIP/5060-b709f998", "DIAL_TRUNK_OPTIONS=tr") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/5060-b709f998", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing Set("SIP/5060-b709f998", "GROUP()=OUT_10") in new stack
-- Executing Macro("SIP/5060-b709f998", "user-callerid|SKIPTTL") in new stack
-- Executing NoOp("SIP/5060-b709f998", "user-callerid: 9006") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/5060-b709f998", "0?report") in new stack
-- Executing GotoIf("SIP/5060-b709f998", "0?start") in new stack
-- Executing Set("SIP/5060-b709f998", "REALCALLERIDNUM=9006") in new stack
[Kasterisk1*CLI>
-- Executing NoOp("SIP/5060-b709f998", "REALCALLERIDNUM is 9006") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b709f998", "AMPUSER=") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b709f998", "AMPUSERCIDNAME=") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/5060-b709f998", "1?report") in new stack
-- Goto (macro-user-callerid,s,11)
-- Executing NoOp("SIP/5060-b709f998", "TTL: ARG1: SKIPTTL") in new stack
-- Executing GotoIf("SIP/5060-b709f998", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/5060-b709f998", "Using CallerID "" <9006>") in new stack
[Kasterisk1*CLI>
-- Executing Macro("SIP/5060-b709f998", "record-enable|9006|OUT") in new stack
-- Executing GotoIf("SIP/5060-b709f998", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("SIP/5060-b709f998", "recordingcheck|20071109-015930|1194573570.135440") in new stack
[Kasterisk1*CLI>
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
[Kasterisk1*CLI>
-- SIP/Addpac-08933248 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Goto("SIP/66.246.197.20-b70257d8", "s-CONGESTION|1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing NoOp("SIP/66.246.197.20-b70257d8", "Dial failed due to CONGESTION - failing through to other trunks") in new stack
-- Executing Macro("SIP/66.246.197.20-b70257d8", "outisbusy|") in new stack
-- Executing Playback("SIP/66.246.197.20-b70257d8", "all-circuits-busy-now|noanswer") in new stack
-- Playing 'all-circuits-busy-now' (language 'en')
[Kasterisk1*CLI>
recordingcheck|20071109-015930|1194573570.135440: No AMPUSER db entry for 9006. Not recording
[Kasterisk1*CLI>
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/5060-b709f998", "No recording needed") in new stack
-- Executing GotoIf("SIP/5060-b709f998", "0?skipoutcid") in new stack
-- Executing Set("SIP/5060-b709f998", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing Macro("SIP/5060-b709f998", "outbound-callerid|10") in new stack
-- Executing GotoIf("SIP/5060-b709f998", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
[Kasterisk1*CLI>
-- Executing NoOp("SIP/5060-b709f998", "REALCALLERIDNUM is 9006") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/5060-b709f998", "1?normcid") in new stack
[Kasterisk1*CLI>
-- Goto (macro-outbound-callerid,s,9)
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b709f998", "USEROUTCID=") in new stack
-- Executing Set("SIP/5060-b709f998", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/5060-b709f998", "TRUNKOUTCID=") in new stack
-- Executing GotoIf("SIP/5060-b709f998", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing GotoIf("SIP/5060-b709f998", "1?usercid") in new stack
-- Goto (macro-outbound-callerid,s,18)
-- Executing GotoIf("SIP/5060-b709f998", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing NoOp("SIP/5060-b709f998", "CallerID set to "" <9006>") in new stack
-- Executing GotoIf("SIP/5060-b709f998", "0?nomax") in new stack
-- Executing GotoIf("SIP/5060-b709f998", "0?chanfull") in new stack
-- Executing DeadAGI("SIP/5060-b709f998", "fixlocalprefix") in new stack
[Kasterisk1*CLI>
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
[Kasterisk1*CLI>
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/5060-b709f998", "OUTNUM=9086236547") in new stack
-- Executing Set("SIP/5060-b709f998", "custom=IAX2/GTT_No_VPN") in new stack
-- Executing GotoIf("SIP/5060-b709f998", "0?customtrunk") in new stack
-- Executing Set("SIP/5060-b709f998", "CDR(userfield)=204.14.9.50") in new stack
-- Executing Dial("SIP/5060-b709f998", "IAX2/GTT_No_VPN/9086236547|300|") in new stack
-- Called GTT_No_VPN/9086236547
[Kasterisk1*CLI>
-- Executing Macro("SIP/5060-b720cfe0", "dialout-trunk|10|9086174443||") in new stack
-- Executing Set("SIP/5060-b720cfe0", "CDR(userfield)=70.42.39.67") in new stack
-- Executing Set("SIP/5060-b720cfe0", "DIAL_TRUNK=10") in new stack
-- Executing Set("SIP/5060-b720cfe0", "_NODEST=") in new stack
-- Executing Set("SIP/5060-b720cfe0", "DIAL_NUMBER=9086174443") in new stack
-- Executing Set("SIP/5060-b720cfe0", "ROUTE_PASSWD=") in new stack
-- Executing Set("SIP/5060-b720cfe0", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing GotoIf("SIP/5060-b720cfe0", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing Set("SIP/5060-b720cfe0", "GROUP()=OUT_10") in new stack
[Kasterisk1*CLI>
-- Executing Macro("SIP/5060-b720cfe0", "user-callerid|SKIPTTL") in new stack
[Kasterisk1*CLI>
-- Executing NoOp("SIP/5060-b720cfe0", "user-callerid: 9010") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/5060-b720cfe0", "0?report") in new stack
-- Executing GotoIf("SIP/5060-b720cfe0", "0?start") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b720cfe0", "REALCALLERIDNUM=9010") in new stack
[Kasterisk1*CLI>
-- Executing NoOp("SIP/5060-b720cfe0", "REALCALLERIDNUM is 9010") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b720cfe0", "AMPUSER=") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/5060-b720cfe0", "AMPUSERCIDNAME=") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/5060-b720cfe0", "1?report") in new stack
-- Goto (macro-user-callerid,s,11)
[Kasterisk1*CLI>
-- Executing NoOp("SIP/5060-b720cfe0", "TTL: ARG1: SKIPTTL") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/5060-b720cfe0", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
[Kasterisk1*CLI>
-- Executing NoOp("SIP/5060-b720cfe0", "Using CallerID "" <9010>") in new stack
[Kasterisk1*CLI>
-- Executing Macro("SIP/5060-b720cfe0", "record-enable|9010|OUT") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/5060-b720cfe0", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("SIP/5060-b720cfe0", "recordingcheck|20071109-015930|1194573570.135442") in new stack
[Kasterisk1*CLI>
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
[Kasterisk1*CLI>
recordingcheck|20071109-015930|1194573570.135442: No AMPUSER db entry for 9010. Not recording
[Kasterisk1*CLI>
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/5060-b720cfe0", "No recording needed") in new stack
-- Executing GotoIf("SIP/5060-b720cfe0", "0?skipoutcid") in new stack
-- Executing Set("SIP/5060-b720cfe0", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing Macro("SIP/5060-b720cfe0", "outbound-callerid|10") in new stack
-- Executing GotoIf("SIP/5060-b720cfe0", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing NoOp("SIP/5060-b720cfe0", "REALCALLERIDNUM is 9010") in new stack
-- Executing GotoIf("SIP/5060-b720cfe0", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing Set("SIP/5060-b720cfe0", "USEROUTCID=") in new stack
-- Executing Set("SIP/5060-b720cfe0", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/5060-b720cfe0", "TRUNKOUTCID=") in new stack
-- Executing GotoIf("SIP/5060-b720cfe0", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing GotoIf("SIP/5060-b720cfe0", "1?usercid") in new stack
-- Goto (macro-outbound-callerid,s,18)
-- Executing GotoIf("SIP/5060-b720cfe0", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing NoOp("SIP/5060-b720cfe0", "CallerID set to "" <9010>") in new stack
-- Executing GotoIf("SIP/5060-b720cfe0", "0?nomax") in new stack
-- Executing GotoIf("SIP/5060-b720cfe0", "0?chanfull") in new stack
-- Executing DeadAGI("SIP/5060-b720cfe0", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
[Kasterisk1*CLI>
-- IAX2/GTT_No_VPN-16419 stopped sounds
-- IAX2/GTT_No_VPN-16419 answered SIP/5060-b70a25b8
[Kasterisk1*CLI>
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/5060-b720cfe0", "OUTNUM=9086174443") in new stack
-- Executing Set("SIP/5060-b720cfe0", "custom=IAX2/GTT_No_VPN") in new stack
-- Executing GotoIf("SIP/5060-b720cfe0", "0?customtrunk") in new stack
-- Executing Set("SIP/5060-b720cfe0", "CDR(userfield)=70.42.39.67") in new stack
-- Executing Dial("SIP/5060-b720cfe0", "IAX2/GTT_No_VPN/9086174443|300|") in new stack
-- Called GTT_No_VPN/9086174443
[Kasterisk1*CLI>
-- Call accepted by 190.80.53.168 (format g723)
-- Format for call is g723
[Kasterisk1*CLI>
-- Call accepted by 190.80.53.168 (format g723)
-- Format for call is g723
[Kasterisk1*CLI>
-- Remote UNIX connection
[Kasterisk1*CLI>
-- Remote UNIX connection disconnected
[Kasterisk1*CLI>
-- Executing Macro("SIP/5060-b7243938", "dialout-trunk|10|9082180795||") in new stack
-- Executing Set("SIP/5060-b7243938", "CDR(userfield)=70.42.39.67") in new stack
-- Executing Set("SIP/5060-b7243938", "DIAL_TRUNK=10") in new stack
-- Executing Set("SIP/5060-b7243938", "_NODEST=") in new stack
-- Executing Set("SIP/5060-b7243938", "DIAL_NUMBER=9082180795") in new stack
-- Executing Set("SIP/5060-b7243938", "ROUTE_PASSWD=") in new stack
-- Executing Set("SIP/5060-b7243938", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing GotoIf("SIP/5060-b7243938", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing Set("SIP/5060-b7243938", "GROUP()=OUT_10") in new stack
-- Executing Macro("SIP/5060-b7243938", "user-callerid|SKIPTTL") in new stack
-- Executing NoOp("SIP/5060-b7243938", "user-callerid: 9010") in new stack
-- Executing GotoIf("SIP/5060-b7243938", "0?report") in new stack
-- Executing GotoIf("SIP/5060-b7243938", "0?start") in new stack
-- Executing Set("SIP/5060-b7243938", "REALCALLERIDNUM=9010") in new stack
-- Executing NoOp("SIP/5060-b7243938", "REALCALLERIDNUM is 9010") in new stack
-- Executing Set("SIP/5060-b7243938", "AMPUSER=") in new stack
-- Executing Set("SIP/5060-b7243938", "AMPUSERCIDNAME=") in new stack
-- Executing GotoIf("SIP/5060-b7243938", "1?report") in new stack
-- Goto (macro-user-callerid,s,11)
-- Executing NoOp("SIP/5060-b7243938", "TTL: ARG1: SKIPTTL") in new stack
-- Executing GotoIf("SIP/5060-b7243938", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/5060-b7243938", "Using CallerID "" <9010>") in new stack
-- Executing Macro("SIP/5060-b7243938", "record-enable|9010|OUT") in new stack
-- Executing GotoIf("SIP/5060-b7243938", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("SIP/5060-b7243938", "recordingcheck|20071109-015932|1194573572.135444") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
[Kasterisk1*CLI>
recordingcheck|20071109-015932|1194573572.135444: No AMPUSER db entry for 9010. Not recording
[Kasterisk1*CLI>
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/5060-b7243938", "No recording needed") in new stack
-- Executing GotoIf("SIP/5060-b7243938", "0?skipoutcid") in new stack
-- Executing Set("SIP/5060-b7243938", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing Macro("SIP/5060-b7243938", "outbound-callerid|10") in new stack
-- Executing GotoIf("SIP/5060-b7243938", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing NoOp("SIP/5060-b7243938", "REALCALLERIDNUM is 9010") in new stack
-- Executing GotoIf("SIP/5060-b7243938", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing Set("SIP/5060-b7243938", "USEROUTCID=") in new stack
-- Executing Set("SIP/5060-b7243938", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/5060-b7243938", "TRUNKOUTCID=") in new stack
-- Executing GotoIf("SIP/5060-b7243938", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing GotoIf("SIP/5060-b7243938", "1?usercid") in new stack
-- Goto (macro-outbound-callerid,s,18)
-- Executing GotoIf("SIP/5060-b7243938", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing NoOp("SIP/5060-b7243938", "CallerID set to "" <9010>") in new stack
-- Executing GotoIf("SIP/5060-b7243938", "0?nomax") in new stack
-- Executing GotoIf("SIP/5060-b7243938", "0?chanfull") in new stack
-- Executing DeadAGI("SIP/5060-b7243938", "fixlocalprefix") in new stack
[Kasterisk1*CLI>
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
[Kasterisk1*CLI>
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/5060-b7243938", "OUTNUM=9082180795") in new stack
-- Executing Set("SIP/5060-b7243938", "custom=IAX2/GTT_No_VPN") in new stack
-- Executing GotoIf("SIP/5060-b7243938", "0?customtrunk") in new stack
-- Executing Set("SIP/5060-b7243938", "CDR(userfield)=70.42.39.67") in new stack
-- Executing Dial("SIP/5060-b7243938", "IAX2/GTT_No_VPN/9082180795|300|") in new stack
-- Called GTT_No_VPN/9082180795
[Kasterisk1*CLI>
-- Executing Playback("SIP/66.246.197.20-b7083060", "pls-try-call-later|noanswer") in new stack
-- Playing 'pls-try-call-later' (language 'en')
[Kasterisk1*CLI>
-- Executing Playback("SIP/66.246.197.20-b70257d8", "pls-try-call-later|noanswer") in new stack
-- Playing 'pls-try-call-later' (language 'en')
[Kasterisk1*CLI>
-- Call accepted by 190.80.53.168 (format g723)
-- Format for call is g723
[Kasterisk1*CLI>
-- Executing Set("SIP/66.246.197.20-b7083060", "CDR(userfield)=66.246.197.20") in new stack
-- Executing Macro("SIP/66.246.197.20-b7083060", "hangupcall") in new stack
-- Executing ResetCDR("SIP/66.246.197.20-b7083060", "w") in new stack
[Kasterisk1*CLI>
-- Executing NoCDR("SIP/66.246.197.20-b7083060", "") in new stack
-- Executing GotoIf("SIP/66.246.197.20-b7083060", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing GotoIf("SIP/66.246.197.20-b7083060", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing Wait("SIP/66.246.197.20-b7083060", "5") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/66.246.197.20-b70257d8", "CDR(userfield)=66.246.197.20") in new stack
-- Executing Macro("SIP/66.246.197.20-b70257d8", "hangupcall") in new stack
-- Executing ResetCDR("SIP/66.246.197.20-b70257d8", "w") in new stack
-- Executing NoCDR("SIP/66.246.197.20-b70257d8", "") in new stack
-- Executing GotoIf("SIP/66.246.197.20-b70257d8", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing GotoIf("SIP/66.246.197.20-b70257d8", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing Wait("SIP/66.246.197.20-b70257d8", "5") in new stack
[Kasterisk1*CLI>
-- Executing Macro("SIP/5060-b707ae28", "dialout-trunk|10|9086549575||") in new stack
-- Executing Set("SIP/5060-b707ae28", "CDR(userfield)=204.14.9.50") in new stack
-- Executing Set("SIP/5060-b707ae28", "DIAL_TRUNK=10") in new stack
-- Executing Set("SIP/5060-b707ae28", "_NODEST=") in new stack
-- Executing Set("SIP/5060-b707ae28", "DIAL_NUMBER=9086549575") in new stack
-- Executing Set("SIP/5060-b707ae28", "ROUTE_PASSWD=") in new stack
-- Executing Set("SIP/5060-b707ae28", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing GotoIf("SIP/5060-b707ae28", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing Set("SIP/5060-b707ae28", "GROUP()=OUT_10") in new stack
-- Executing Macro("SIP/5060-b707ae28", "user-callerid|SKIPTTL") in new stack
-- Executing NoOp("SIP/5060-b707ae28", "user-callerid: 9006") in new stack
-- Executing GotoIf("SIP/5060-b707ae28", "0?report") in new stack
-- Executing GotoIf("SIP/5060-b707ae28", "0?start") in new stack
-- Executing Set("SIP/5060-b707ae28", "REALCALLERIDNUM=9006") in new stack
-- Executing NoOp("SIP/5060-b707ae28", "REALCALLERIDNUM is 9006") in new stack
-- Executing Set("SIP/5060-b707ae28", "AMPUSER=") in new stack
-- Executing Set("SIP/5060-b707ae28", "AMPUSERCIDNAME=") in new stack
-- Executing GotoIf("SIP/5060-b707ae28", "1?report") in new stack
-- Goto (macro-user-callerid,s,11)
-- Executing NoOp("SIP/5060-b707ae28", "TTL: ARG1: SKIPTTL") in new stack
-- Executing GotoIf("SIP/5060-b707ae28", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/5060-b707ae28", "Using CallerID "" <9006>") in new stack
-- Executing Macro("SIP/5060-b707ae28", "record-enable|9006|OUT") in new stack
-- Executing GotoIf("SIP/5060-b707ae28", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("SIP/5060-b707ae28", "recordingcheck|20071109-015936|1194573576.135446") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
[Kasterisk1*CLI>
recordingcheck|20071109-015936|1194573576.135446: No AMPUSER db entry for 9006. Not recording
[Kasterisk1*CLI>
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/5060-b707ae28", "No recording needed") in new stack
-- Executing GotoIf("SIP/5060-b707ae28", "0?skipoutcid") in new stack
-- Executing Set("SIP/5060-b707ae28", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing Macro("SIP/5060-b707ae28", "outbound-callerid|10") in new stack
-- Executing GotoIf("SIP/5060-b707ae28", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing NoOp("SIP/5060-b707ae28", "REALCALLERIDNUM is 9006") in new stack
-- Executing GotoIf("SIP/5060-b707ae28", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing Set("SIP/5060-b707ae28", "USEROUTCID=") in new stack
-- Executing Set("SIP/5060-b707ae28", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/5060-b707ae28", "TRUNKOUTCID=") in new stack
-- Executing GotoIf("SIP/5060-b707ae28", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing GotoIf("SIP/5060-b707ae28", "1?usercid") in new stack
-- Goto (macro-outbound-callerid,s,18)
-- Executing GotoIf("SIP/5060-b707ae28", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing NoOp("SIP/5060-b707ae28", "CallerID set to "" <9006>") in new stack
-- Executing GotoIf("SIP/5060-b707ae28", "0?nomax") in new stack
-- Executing GotoIf("SIP/5060-b707ae28", "0?chanfull") in new stack
-- Executing DeadAGI("SIP/5060-b707ae28", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
[Kasterisk1*CLI>
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/5060-b707ae28", "OUTNUM=9086549575") in new stack
-- Executing Set("SIP/5060-b707ae28", "custom=IAX2/GTT_No_VPN") in new stack
-- Executing GotoIf("SIP/5060-b707ae28", "0?customtrunk") in new stack
-- Executing Set("SIP/5060-b707ae28", "CDR(userfield)=204.14.9.50") in new stack
-- Executing Dial("SIP/5060-b707ae28", "IAX2/GTT_No_VPN/9086549575|300|") in new stack
-- Called GTT_No_VPN/9086549575
[Kasterisk1*CLI>
-- Call accepted by 190.80.53.168 (format g723)
-- Format for call is g723
[Kasterisk1*CLI>
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/66.246.197.20-b7083060' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/66.246.197.20-b7083060' in macro 'outisbusy'
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/66.246.197.20-b7083060'
[Kasterisk1*CLI>
== End MixMonitor Recording SIP/66.246.197.20-b7083060
[Kasterisk1*CLI>
-- Hungup 'IAX2/Adesh_Trinidad-16423'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/5060-b72e2d80' in macro 'dialout-trunk'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/5060-b72e2d80'
-- Executing Macro("SIP/5060-b72e2d80", "hangupcall") in new stack
-- Executing ResetCDR("SIP/5060-b72e2d80", "w") in new stack
[Kasterisk1*CLI>
-- Executing NoCDR("SIP/5060-b72e2d80", "") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/5060-b72e2d80", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing GotoIf("SIP/5060-b72e2d80", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing Wait("SIP/5060-b72e2d80", "5") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/5060-b72e2d80' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/5060-b72e2d80'
[Kasterisk1*CLI>
-- Hungup 'IAX2/GTT_No_VPN-16431'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/5060-b7243938' in macro 'dialout-trunk'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/5060-b7243938'
-- Executing Macro("SIP/5060-b7243938", "hangupcall") in new stack
-- Executing ResetCDR("SIP/5060-b7243938", "w") in new stack
[Kasterisk1*CLI>
-- Executing NoCDR("SIP/5060-b7243938", "") in new stack
-- Executing GotoIf("SIP/5060-b7243938", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing GotoIf("SIP/5060-b7243938", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing Wait("SIP/5060-b7243938", "5") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/5060-b7243938' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/5060-b7243938'
[Kasterisk1*CLI>
-- Executing Hangup("SIP/66.246.197.20-b70257d8", "") in new stack
== Spawn extension (macro-hangupcall, s, 10) exited non-zero on 'SIP/66.246.197.20-b70257d8' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 10) exited non-zero on 'SIP/66.246.197.20-b70257d8' in macro 'outisbusy'
== Spawn extension (macro-hangupcall, s, 10) exited non-zero on 'SIP/66.246.197.20-b70257d8'
== End MixMonitor Recording SIP/66.246.197.20-b70257d8
[Kasterisk1*CLI>
-- IAX2/GTT_No_VPN-16429 is ringing
[Kasterisk1*CLI>
Sending to 64.192.196.12 : 5060 (non-NAT)
Found RTP audio format 18
Found RTP audio format 19
Peer audio RTP is at port 64.192.196.12:19166
Found description format G729
Found description format CN
Capabilities: us - 0x8070f (g723|gsm|ulaw|alaw|g729|speex|ilbc|h263), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x2 (CN), combined - 0x0 (nothing)
Looking for 18686398571 in from-sip-external (domain 208.75.208.90)
list_route: hop: <sip:64.192.196.12:5060>
Transmitting (no NAT) to 64.192.196.12:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 64.192.196.12:5060;branch=z9hG4bK101310;received=64.192.196.12
From: <sip:64.192.196.12>;tag=2A03BAB5-2F6
To: <sip:18686398571@208.75.208.90>
Call-ID: 4D42D695-324711D6-98DBF52C-1F15DBCD@64.192.196.12
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:18686398571@208.75.208.90>
Content-Length: 0
---
-- Executing NoOp("SIP/64.192.196.12-b7243938", "Received incoming SIP connection from unknown peer to 18686398571") in new stack
-- Executing Set("SIP/64.192.196.12-b7243938", "DID=18686398571") in new stack
-- Executing Goto("SIP/64.192.196.12-b7243938", "s|1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing Ringing("SIP/64.192.196.12-b7243938", "") in new stack
[Kasterisk1*CLI>
Transmitting (no NAT) to 64.192.196.12:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 64.192.196.12:5060;branch=z9hG4bK101310;received=64.192.196.12
From: <sip:64.192.196.12>;tag=2A03BAB5-2F6
To: <sip:18686398571@208.75.208.90>;tag=as136fa5a2
Call-ID: 4D42D695-324711D6-98DBF52C-1F15DBCD@64.192.196.12
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:18686398571@208.75.208.90>
Content-Length: 0
---
-- Executing Set("SIP/64.192.196.12-b7243938", "CDR(userfield)=64.192.196.12") in new stack
-- Executing GotoIf("SIP/64.192.196.12-b7243938", "1?from-trunk|18686398571|1") in new stack
-- Goto (from-trunk,18686398571,1)
Scheduling destruction of call '4D42D695-324711D6-98DBF52C-1F15DBCD@64.192.196.12' in 32000 ms
Reliably Transmitting (no NAT) to 64.192.196.12:5060:
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 64.192.196.12:5060;branch=z9hG4bK101310;received=64.192.196.12
From: <sip:64.192.196.12>;tag=2A03BAB5-2F6
To: <sip:18686398571@208.75.208.90>;tag=as136fa5a2
Call-ID: 4D42D695-324711D6-98DBF52C-1F15DBCD@64.192.196.12
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:18686398571@208.75.208.90>
Content-Length: 0
---
[Kasterisk1*CLI>
<-- SIP read from 64.192.196.12:5060:
ACK sip:18686398571@208.75.208.90:5060 SIP/2.0
Via: SIP/2.0/UDP 64.192.196.12:5060;branch=z9hG4bK101310
From: <sip:64.192.196.12>;tag=2A03BAB5-2F6
To: <sip:18686398571@208.75.208.90>;tag=as136fa5a2
Date: Sat, 09 Mar 2002 03:48:07 GMT
Call-ID: 4D42D695-324711D6-98DBF52C-1F15DBCD@64.192.196.12
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
--- (10 headers 0 lines) ---
[Kasterisk1*CLI>
-- Hungup 'IAX2/GTT_No_VPN-16390'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/5060-b720cfe0' in macro 'dialout-trunk'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/5060-b720cfe0'
-- Executing Macro("SIP/5060-b720cfe0", "hangupcall") in new stack
-- Executing ResetCDR("SIP/5060-b720cfe0", "w") in new stack
[Kasterisk1*CLI>
-- Executing NoCDR("SIP/5060-b720cfe0", "") in new stack
-- Executing GotoIf("SIP/5060-b720cfe0", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing GotoIf("SIP/5060-b720cfe0", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing Wait("SIP/5060-b720cfe0", "5") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/5060-b720cfe0' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/5060-b720cfe0'
[Kasterisk1*CLI>
-- Hungup 'IAX2/GTT_No_VPN-16438'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/5060-b7267be8' in macro 'dialout-trunk'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/5060-b7267be8'
-- Executing Macro("SIP/5060-b7267be8", "hangupcall") in new stack
-- Executing ResetCDR("SIP/5060-b7267be8", "w") in new stack
-- Executing NoCDR("SIP/5060-b7267be8", "") in new stack
-- Executing GotoIf("SIP/5060-b7267be8", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing GotoIf("SIP/5060-b7267be8", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing Wait("SIP/5060-b7267be8", "5") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/5060-b7267be8' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/5060-b7267be8'
[Kasterisk1*CLI>
-- IAX2/GTT_No_VPN-16415 stopped sounds
-- IAX2/GTT_No_VPN-16415 answered SIP/5060-b72a8b70
[Kasterisk1*CLI>
-- Executing Macro("SIP/5060-b729ec30", "dialout-trunk|10|9086442490||") in new stack
-- Executing Set("SIP/5060-b729ec30", "CDR(userfield)=70.42.39.67") in new stack
-- Executing Set("SIP/5060-b729ec30", "DIAL_TRUNK=10") in new stack
-- Executing Set("SIP/5060-b729ec30", "_NODEST=") in new stack
-- Executing Set("SIP/5060-b729ec30", "DIAL_NUMBER=9086442490") in new stack
-- Executing Set("SIP/5060-b729ec30", "ROUTE_PASSWD=") in new stack
-- Executing Set("SIP/5060-b729ec30", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing GotoIf("SIP/5060-b729ec30", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing Set("SIP/5060-b729ec30", "GROUP()=OUT_10") in new stack
-- Executing Macro("SIP/5060-b729ec30", "user-callerid|SKIPTTL") in new stack
-- Executing NoOp("SIP/5060-b729ec30", "user-callerid: 9010") in new stack
-- Executing GotoIf("SIP/5060-b729ec30", "0?report") in new stack
-- Executing GotoIf("SIP/5060-b729ec30", "0?start") in new stack
-- Executing Set("SIP/5060-b729ec30", "REALCALLERIDNUM=9010") in new stack
-- Executing NoOp("SIP/5060-b729ec30", "REALCALLERIDNUM is 9010") in new stack
-- Executing Set("SIP/5060-b729ec30", "AMPUSER=") in new stack
-- Executing Set("SIP/5060-b729ec30", "AMPUSERCIDNAME=") in new stack
-- Executing GotoIf("SIP/5060-b729ec30", "1?report") in new stack
-- Goto (macro-user-callerid,s,11)
-- Executing NoOp("SIP/5060-b729ec30", "TTL: ARG1: SKIPTTL") in new stack
-- Executing GotoIf("SIP/5060-b729ec30", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/5060-b729ec30", "Using CallerID "" <9010>") in new stack
-- Executing Macro("SIP/5060-b729ec30", "record-enable|9010|OUT") in new stack
-- Executing GotoIf("SIP/5060-b729ec30", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("SIP/5060-b729ec30", "recordingcheck|20071109-015950|1194573590.135449") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
[Kasterisk1*CLI>
recordingcheck|20071109-015950|1194573590.135449: No AMPUSER db entry for 9010. Not recording
[Kasterisk1*CLI>
-- AGI Script recordingcheck completed, returning 0
[Kasterisk1*CLI>
-- Executing NoOp("SIP/5060-b729ec30", "No recording needed") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/5060-b729ec30", "0?skipoutcid") in new stack
-- Executing Set("SIP/5060-b729ec30", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing Macro("SIP/5060-b729ec30", "outbound-callerid|10") in new stack
-- Executing GotoIf("SIP/5060-b729ec30", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing NoOp("SIP/5060-b729ec30", "REALCALLERIDNUM is 9010") in new stack
-- Executing GotoIf("SIP/5060-b729ec30", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing Set("SIP/5060-b729ec30", "USEROUTCID=") in new stack
-- Executing Set("SIP/5060-b729ec30", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/5060-b729ec30", "TRUNKOUTCID=") in new stack
-- Executing GotoIf("SIP/5060-b729ec30", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing GotoIf("SIP/5060-b729ec30", "1?usercid") in new stack
-- Goto (macro-outbound-callerid,s,18)
-- Executing GotoIf("SIP/5060-b729ec30", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing NoOp("SIP/5060-b729ec30", "CallerID set to "" <9010>") in new stack
-- Executing GotoIf("SIP/5060-b729ec30", "0?nomax") in new stack
-- Executing GotoIf("SIP/5060-b729ec30", "0?chanfull") in new stack
-- Executing DeadAGI("SIP/5060-b729ec30", "fixlocalprefix") in new stack
[Kasterisk1*CLI>
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
[Kasterisk1*CLI>
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/5060-b729ec30", "OUTNUM=9086442490") in new stack
-- Executing Set("SIP/5060-b729ec30", "custom=IAX2/GTT_No_VPN") in new stack
-- Executing GotoIf("SIP/5060-b729ec30", "0?customtrunk") in new stack
-- Executing Set("SIP/5060-b729ec30", "CDR(userfield)=70.42.39.67") in new stack
-- Executing Dial("SIP/5060-b729ec30", "IAX2/GTT_No_VPN/9086442490|300|") in new stack
-- Called GTT_No_VPN/9086442490
[Kasterisk1*CLI>
-- Call accepted by 190.80.53.168 (format g723)
-- Format for call is g723
[Kasterisk1*CLI>
-- Executing Macro("SIP/5060-b720cfd0", "dialout-trunk|10|9086435737||") in new stack
-- Executing Set("SIP/5060-b720cfd0", "CDR(userfield)=204.14.9.50") in new stack
-- Executing Set("SIP/5060-b720cfd0", "DIAL_TRUNK=10") in new stack
-- Executing Set("SIP/5060-b720cfd0", "_NODEST=") in new stack
-- Executing Set("SIP/5060-b720cfd0", "DIAL_NUMBER=9086435737") in new stack
-- Executing Set("SIP/5060-b720cfd0", "ROUTE_PASSWD=") in new stack
-- Executing Set("SIP/5060-b720cfd0", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing GotoIf("SIP/5060-b720cfd0", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing Set("SIP/5060-b720cfd0", "GROUP()=OUT_10") in new stack
-- Executing Macro("SIP/5060-b720cfd0", "user-callerid|SKIPTTL") in new stack
-- Executing NoOp("SIP/5060-b720cfd0", "user-callerid: 9006") in new stack
-- Executing GotoIf("SIP/5060-b720cfd0", "0?report") in new stack
-- Executing GotoIf("SIP/5060-b720cfd0", "0?start") in new stack
-- Executing Set("SIP/5060-b720cfd0", "REALCALLERIDNUM=9006") in new stack
-- Executing NoOp("SIP/5060-b720cfd0", "REALCALLERIDNUM is 9006") in new stack
-- Executing Set("SIP/5060-b720cfd0", "AMPUSER=") in new stack
-- Executing Set("SIP/5060-b720cfd0", "AMPUSERCIDNAME=") in new stack
-- Executing GotoIf("SIP/5060-b720cfd0", "1?report") in new stack
-- Goto (macro-user-callerid,s,11)
-- Executing NoOp("SIP/5060-b720cfd0", "TTL: ARG1: SKIPTTL") in new stack
-- Executing GotoIf("SIP/5060-b720cfd0", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/5060-b720cfd0", "Using CallerID "" <9006>") in new stack
-- Executing Macro("SIP/5060-b720cfd0", "record-enable|9006|OUT") in new stack
-- Executing GotoIf("SIP/5060-b720cfd0", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("SIP/5060-b720cfd0", "recordingcheck|20071109-015952|1194573908.135451") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
[Kasterisk1*CLI>
recordingcheck|20071109-015952|1194573908.135451: No AMPUSER db entry for 9006. Not recording
[Kasterisk1*CLI>
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/5060-b720cfd0", "No recording needed") in new stack
-- Executing GotoIf("SIP/5060-b720cfd0", "0?skipoutcid") in new stack
-- Executing Set("SIP/5060-b720cfd0", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing Macro("SIP/5060-b720cfd0", "outbound-callerid|10") in new stack
-- Executing GotoIf("SIP/5060-b720cfd0", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing NoOp("SIP/5060-b720cfd0", "REALCALLERIDNUM is 9006") in new stack
-- Executing GotoIf("SIP/5060-b720cfd0", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing Set("SIP/5060-b720cfd0", "USEROUTCID=") in new stack
-- Executing Set("SIP/5060-b720cfd0", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/5060-b720cfd0", "TRUNKOUTCID=") in new stack
-- Executing GotoIf("SIP/5060-b720cfd0", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing GotoIf("SIP/5060-b720cfd0", "1?usercid") in new stack
-- Goto (macro-outbound-callerid,s,18)
-- Executing GotoIf("SIP/5060-b720cfd0", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing NoOp("SIP/5060-b720cfd0", "CallerID set to "" <9006>") in new stack
-- Executing GotoIf("SIP/5060-b720cfd0", "0?nomax") in new stack
-- Executing GotoIf("SIP/5060-b720cfd0", "0?chanfull") in new stack
-- Executing DeadAGI("SIP/5060-b720cfd0", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
[Kasterisk1*CLI>
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/5060-b720cfd0", "OUTNUM=9086435737") in new stack
-- Executing Set("SIP/5060-b720cfd0", "custom=IAX2/GTT_No_VPN") in new stack
-- Executing GotoIf("SIP/5060-b720cfd0", "0?customtrunk") in new stack
-- Executing Set("SIP/5060-b720cfd0", "CDR(userfield)=204.14.9.50") in new stack
-- Executing Dial("SIP/5060-b720cfd0", "IAX2/GTT_No_VPN/9086435737|300|") in new stack
-- Called GTT_No_VPN/9086435737
[Kasterisk1*CLI>
-- Call accepted by 190.80.53.168 (format g723)
-- Format for call is g723
[Kasterisk1*CLI>
-- Hungup 'IAX2/GTT_No_VPN-16389'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/5060-b709f998' in macro 'dialout-trunk'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/5060-b709f998'
-- Executing Macro("SIP/5060-b709f998", "hangupcall") in new stack
-- Executing ResetCDR("SIP/5060-b709f998", "w") in new stack
[Kasterisk1*CLI>
-- Executing NoCDR("SIP/5060-b709f998", "") in new stack
-- Executing GotoIf("SIP/5060-b709f998", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing GotoIf("SIP/5060-b709f998", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing Wait("SIP/5060-b709f998", "5") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/5060-b709f998' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/5060-b709f998'
[Kasterisk1*CLI>
-- IAX2/GTT_No_VPN-16429 stopped sounds
[Kasterisk1*CLI>
-- IAX2/GTT_No_VPN-16429 answered SIP/5060-b7219088
[Kasterisk1*CLI>
-- Executing Macro("SIP/64.15.135.252-b709f998", "dialout-trunk|10|9086528122||") in new stack
-- Executing Set("SIP/64.15.135.252-b709f998", "CDR(userfield)=64.15.135.252") in new stack
-- Executing Set("SIP/64.15.135.252-b709f998", "DIAL_TRUNK=10") in new stack
-- Executing Set("SIP/64.15.135.252-b709f998", "_NODEST=") in new stack
-- Executing Set("SIP/64.15.135.252-b709f998", "DIAL_NUMBER=9086528122") in new stack
-- Executing Set("SIP/64.15.135.252-b709f998", "ROUTE_PASSWD=") in new stack
-- Executing Set("SIP/64.15.135.252-b709f998", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b709f998", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing Set("SIP/64.15.135.252-b709f998", "GROUP()=OUT_10") in new stack
-- Executing Macro("SIP/64.15.135.252-b709f998", "user-callerid|SKIPTTL") in new stack
-- Executing NoOp("SIP/64.15.135.252-b709f998", "user-callerid: 7aaa 9015") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b709f998", "0?report") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b709f998", "0?start") in new stack
-- Executing Set("SIP/64.15.135.252-b709f998", "REALCALLERIDNUM=9015") in new stack
-- Executing NoOp("SIP/64.15.135.252-b709f998", "REALCALLERIDNUM is 9015") in new stack
-- Executing Set("SIP/64.15.135.252-b709f998", "AMPUSER=") in new stack
-- Executing Set("SIP/64.15.135.252-b709f998", "AMPUSERCIDNAME=") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b709f998", "1?report") in new stack
-- Goto (macro-user-callerid,s,11)
-- Executing NoOp("SIP/64.15.135.252-b709f998", "TTL: ARG1: SKIPTTL") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b709f998", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/64.15.135.252-b709f998", "Using CallerID "7aaa" <9015>") in new stack
-- Executing Macro("SIP/64.15.135.252-b709f998", "record-enable|9015|OUT") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b709f998", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("SIP/64.15.135.252-b709f998", "recordingcheck|20071109-015955|1194573595.135453") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
[Kasterisk1*CLI>
recordingcheck|20071109-015955|1194573595.135453: No AMPUSER db entry for 9015. Not recording
[Kasterisk1*CLI>
-- AGI Script recordingcheck completed, returning 0
[Kasterisk1*CLI>
-- Executing NoOp("SIP/64.15.135.252-b709f998", "No recording needed") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/64.15.135.252-b709f998", "0?skipoutcid") in new stack
[Kasterisk1*CLI>
-- Executing Set("SIP/64.15.135.252-b709f998", "DIAL_TRUNK_OPTIONS=") in new stack
[Kasterisk1*CLI>
-- Executing Macro("SIP/64.15.135.252-b709f998", "outbound-callerid|10") in new stack
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/64.15.135.252-b709f998", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing NoOp("SIP/64.15.135.252-b709f998", "REALCALLERIDNUM is 9015") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b709f998", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
[Kasterisk1*CLI>
-- Executing Set("SIP/64.15.135.252-b709f998", "USEROUTCID=") in new stack
-- Executing Set("SIP/64.15.135.252-b709f998", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/64.15.135.252-b709f998", "TRUNKOUTCID=") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b709f998", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing GotoIf("SIP/64.15.135.252-b709f998", "1?usercid") in new stack
-- Goto (macro-outbound-callerid,s,18)
[Kasterisk1*CLI>
-- Executing GotoIf("SIP/64.15.135.252-b709f998", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
[Kasterisk1*CLI>
-- Executing NoOp("SIP/64.15.135.252-b709f998", "CallerID set to "7aaa" <9015>") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b709f998", "0?nomax") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b709f998", "0?chanfull") in new stack
-- Executing DeadAGI("SIP/64.15.135.252-b709f998", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
[Kasterisk1*CLI>
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/64.15.135.252-b709f998", "OUTNUM=9086528122") in new stack
-- Executing Set("SIP/64.15.135.252-b709f998", "custom=IAX2/GTT_No_VPN") in new stack
-- Executing GotoIf("SIP/64.15.135.252-b709f998", "0?customtrunk") in new stack
-- Executing Set("SIP/64.15.135.252-b709f998", "CDR(userfield)=64.15.135.252") in new stack
-- Executing Dial("SIP/64.15.135.252-b709f998", "IAX2/GTT_No_VPN/9086528122|300|") in new stack
-- Called GTT_No_VPN/9086528122
[Kasterisk1*CLI>
-- Call accepted by 190.80.53.168 (format g723)
-- Format for call is g723
[Kasterisk1*CLI>
Destroying call '4D42D695-324711D6-98DBF52C-1F15DBCD@64.192.196.12'
[Kasterisk1*CLI>
12 headers, 0 lines
Reliably Transmitting (NAT) to 64.192.196.12:5060:
OPTIONS sip:64.192.196.12 SIP/2.0
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK70501f9f;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as339e6699
To: <sip:64.192.196.12>
Contact: <sip:Unknown@208.75.208.90>
Call-ID: 6e44696160a17cf622b8cf0f16e3b7c9@208.75.208.90
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 09 Nov 2007 02:00:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
[Kasterisk1*CLI>
<-- SIP read from 64.192.196.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK70501f9f;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as339e6699
To: <sip:64.192.196.12>;tag=2A044A25-1FE4
Date: Sat, 09 Mar 2002 03:48:44 GMT
Call-ID: 6e44696160a17cf622b8cf0f16e3b7c9@208.75.208.90
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,resource-priority,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 451
v=0
o=CiscoSystemsSIP-GW-UserAgent 7414 8931 IN IP4 64.192.196.12
s=SIP Call
c=IN IP4 64.192.196.12
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 15 3
c=IN IP4 64.192.196.12
m=image 0 udptl t38
c=IN IP4 64.192.196.12
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
--- (14 headers 18 lines) ---
[Kasterisk1*CLI>
Destroying call '6e44696160a17cf622b8cf0f16e3b7c9@208.75.208.90'
[Kasterisk1*CLI>
12 headers, 0 lines
Reliably Transmitting (NAT) to 64.192.196.12:5060:
OPTIONS sip:64.192.196.12 SIP/2.0
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK64b1280b;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as4f054303
To: <sip:64.192.196.12>
Contact: <sip:Unknown@208.75.208.90>
Call-ID: 516ea2494a83f2f25948995c11403501@208.75.208.90
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 09 Nov 2007 02:00:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
[Kasterisk1*CLI>
<-- SIP read from 64.192.196.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK64b1280b;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as4f054303
To: <sip:64.192.196.12>;tag=2A044A5D-1491
Date: Sat, 09 Mar 2002 03:48:44 GMT
Call-ID: 516ea2494a83f2f25948995c11403501@208.75.208.90
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,resource-priority,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 451
v=0
o=CiscoSystemsSIP-GW-UserAgent 5931 8451 IN IP4 64.192.196.12
s=SIP Call
c=IN IP4 64.192.196.12
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 15 3
c=IN IP4 64.192.196.12
m=image 0 udptl t38
c=IN IP4 64.192.196.12
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
--- (14 headers 18 lines) ---
[Kasterisk1*CLI>
Destroying call '516ea2494a83f2f25948995c11403501@208.75.208.90'
[Kasterisk1*CLI>
12 headers, 0 lines
Reliably Transmitting (NAT) to 64.192.196.12:5060:
OPTIONS sip:64.192.196.12 SIP/2.0
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK48c78ef8;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as2f9ecf87
To: <sip:64.192.196.12>
Contact: <sip:Unknown@208.75.208.90>
Call-ID: 731a95c9459687250f842f03444ac03a@208.75.208.90
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 09 Nov 2007 02:01:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
[Kasterisk1*CLI>
<-- SIP read from 64.192.196.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK48c78ef8;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as2f9ecf87
To: <sip:64.192.196.12>;tag=2A0534B3-165C
Date: Sat, 09 Mar 2002 03:49:44 GMT
Call-ID: 731a95c9459687250f842f03444ac03a@208.75.208.90
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,resource-priority,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 451
v=0
o=CiscoSystemsSIP-GW-UserAgent 6830 8072 IN IP4 64.192.196.12
s=SIP Call
c=IN IP4 64.192.196.12
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 15 3
c=IN IP4 64.192.196.12
m=image 0 udptl t38
c=IN IP4 64.192.196.12
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
--- (14 headers 18 lines) ---
[Kasterisk1*CLI>
Destroying call '731a95c9459687250f842f03444ac03a@208.75.208.90'
[Kasterisk1*CLI>
12 headers, 0 lines
Reliably Transmitting (NAT) to 64.192.196.12:5060:
OPTIONS sip:64.192.196.12 SIP/2.0
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK256283ad;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as59c281c0
To: <sip:64.192.196.12>
Contact: <sip:Unknown@208.75.208.90>
Call-ID: 4209dbee20f7ef7c2e20640c307a886a@208.75.208.90
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 09 Nov 2007 02:01:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
[Kasterisk1*CLI>
<-- SIP read from 64.192.196.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK256283ad;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as59c281c0
To: <sip:64.192.196.12>;tag=2A0534EB-64
Date: Sat, 09 Mar 2002 03:49:44 GMT
Call-ID: 4209dbee20f7ef7c2e20640c307a886a@208.75.208.90
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,resource-priority,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 451
v=0
o=CiscoSystemsSIP-GW-UserAgent 5423 5156 IN IP4 64.192.196.12
s=SIP Call
c=IN IP4 64.192.196.12
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 15 3
c=IN IP4 64.192.196.12
m=image 0 udptl t38
c=IN IP4 64.192.196.12
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a
[Kasterisk1*CLI>
=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
--- (14 headers 18 lines) ---
[Kasterisk1*CLI>
Destroying call '4209dbee20f7ef7c2e20640c307a886a@208.75.208.90'
[Kasterisk1*CLI>
12 headers, 0 lines
[Kasterisk1*CLI>
Reliably Transmitting (NAT) to 64.192.196.12:5060:
OPTIONS sip:64.192.196.12 SIP/2.0
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK762af7a4;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as278cf9c8
To: <sip:64.192.196.12>
Contact: <sip:Unknown@208.75.208.90>
Call-ID: 0880b3c1751f020d1d6d296f2be6e8cf@208.75.208.90
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 09 Nov 2007 02:02:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
[Kasterisk1*CLI>
<-- SIP read from 64.192.196.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK762af7a4;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as278cf9c8
To: <sip:64.192.196.12>;tag=2A061F46-17F5
Date: Sat, 09 Mar 2002 03:50:44 GMT
Call-ID: 0880b3c1751f020d1d6d296f2be6e8cf@208.75.208.90
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,resource-priority,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 451
v=0
o=CiscoSystemsSIP-GW-UserAgent 9549 6173 IN IP4 64.192.196.12
s=SIP Call
c=IN IP4 64.192.196.12
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 15 3
c=IN IP4 64.192.196.12
m=image 0 udptl t38
c=IN IP4 64.192.196.12
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
--- (14 headers 18 lines) ---
[Kasterisk1*CLI>
Destroying call '0880b3c1751f020d1d6d296f2be6e8cf@208.75.208.90'
[Kasterisk1*CLI>
12 headers, 0 lines
[Kasterisk1*CLI>
Reliably Transmitting (NAT) to 64.192.196.12:5060:
OPTIONS sip:64.192.196.12 SIP/2.0
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK728b91a3;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as3ed3c5b3
To: <sip:64.192.196.12>
Contact: <sip:Unknown@208.75.208.90>
Call-ID: 2b8854a4447f86d17a86ae09032e0783@208.75.208.90
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 09 Nov 2007 02:02:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
[Kasterisk1*CLI>
<-- SIP read from 64.192.196.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK728b91a3;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as3ed3c5b3
To: <sip:64.192.196.12>;tag=2A061F7A-666
Date: Sat, 09 Mar 2002 03:50:44 GMT
Call-ID: 2b8854a4447f86d17a86ae09032e0783@208.75.208.90
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,resource-priority,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 451
v=0
o=CiscoSystemsSIP-GW-UserAgent 4162 5337 IN IP4 64.192.196.12
s=SIP Call
c=IN IP4 64.192.196.12
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 15 3
c=IN IP4 64.192.196.12
m=image 0 udptl t38
c=IN IP4 64.192.196.12
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
[Kasterisk1*CLI>
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
--- (14 headers 18 lines) ---
[Kasterisk1*CLI>
Destroying call '2b8854a4447f86d17a86ae09032e0783@208.75.208.90'
[Kasterisk1*CLI>
12 headers, 0 lines
[Kasterisk1*CLI>
Reliably Transmitting (NAT) to 64.192.196.12:5060:
OPTIONS sip:64.192.196.12 SIP/2.0
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK37970058;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as163e903a
To: <sip:64.192.196.12>
Contact: <sip:Unknown@208.75.208.90>
Call-ID: 0e37e80f766216740ff523e827f50fa9@208.75.208.90
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 09 Nov 2007 02:03:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
[Kasterisk1*CLI>
<-- SIP read from 64.192.196.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK37970058;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as163e903a
To: <sip:64.192.196.12>;tag=2A0709D4-23D8
Date: Sat, 09 Mar 2002 03:51:44 GMT
Call-ID: 0e37e80f766216740ff523e827f50fa9@208.75.208.90
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,resource-priority,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 451
v=0
o=CiscoSystemsSIP-GW-UserAgent 9014 1486 IN IP4 64.192.196.12
s=SIP Call
c=IN IP4 64.192.196.12
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 15 3
c=IN IP4 64.192.196.12
m=image 0 udptl t38
c=IN IP4 64.192.196.12
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
--- (14 headers 18 lines) ---
[Kasterisk1*CLI>
Destroying call '0e37e80f766216740ff523e827f50fa9@208.75.208.90'
[Kasterisk1*CLI>
12 headers, 0 lines
Reliably Transmitting (NAT) to 64.192.196.12:5060:
OPTIONS sip:64.192.196.12 SIP/2.0
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK564c0132;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as64e62563
To: <sip:64.192.196.12>
Contact: <sip:Unknown@208.75.208.90>
Call-ID: 4dbaebc56d89612e04465b87301594d4@208.75.208.90
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 09 Nov 2007 02:03:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
[Kasterisk1*CLI>
<-- SIP read from 64.192.196.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK564c0132;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as64e62563
To: <sip:64.192.196.12>;tag=2A070A0C-E51
Date: Sat, 09 Mar 2002 03:51:44 GMT
Call-ID: 4dbaebc56d89612e04465b87301594d4@208.75.208.90
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,resource-priority,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 450
v=0
o=CiscoSystemsSIP-GW-UserAgent 467 2287 IN IP4 64.192.196.12
s=SIP Call
c=IN IP4 64.192.196.12
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 15 3
c=IN IP4 64.192.196.12
m=image 0 udptl t38
c=IN IP4 64.192.196.12
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a
[Kasterisk1*CLI>
=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
--- (14 headers 18 lines) ---
[Kasterisk1*CLI>
Destroying call '4dbaebc56d89612e04465b87301594d4@208.75.208.90'
[Kasterisk1*CLI>
12 headers, 0 lines
Reliably Transmitting (NAT) to 64.192.196.12:5060:
OPTIONS sip:64.192.196.12 SIP/2.0
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK3e61e3fa;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as4292f82c
To: <sip:64.192.196.12>
Contact: <sip:Unknown@208.75.208.90>
Call-ID: 3729199e31586dce039113827fb908fc@208.75.208.90
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 09 Nov 2007 02:04:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
[Kasterisk1*CLI>
<-- SIP read from 64.192.196.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK3e61e3fa;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as4292f82c
To: <sip:64.192.196.12>;tag=2A07F463-223A
Date: Sat, 09 Mar 2002 03:52:44 GMT
Call-ID: 3729199e31586dce039113827fb908fc@208.75.208.90
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,resource-priority,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 450
v=0
o=CiscoSystemsSIP-GW-UserAgent 5399 269 IN IP4 64.192.196.12
s=SIP Call
c=IN IP4 64.192.196.12
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 15 3
c=IN IP4 64.192.196.12
m=image 0 udptl t38
c=IN IP4 64.192.196.12
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
[Kasterisk1*CLI>
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
--- (14 headers 18 lines) ---
[Kasterisk1*CLI>
Destroying call '3729199e31586dce039113827fb908fc@208.75.208.90'
[Kasterisk1*CLI>
12 headers, 0 lines
[Kasterisk1*CLI>
Reliably Transmitting (NAT) to 64.192.196.12:5060:
OPTIONS sip:64.192.196.12 SIP/2.0
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK42d89e6f;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as7793e163
To: <sip:64.192.196.12>
Contact: <sip:Unknown@208.75.208.90>
Call-ID: 07dbb55552be22af4750756258b79e06@208.75.208.90
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 09 Nov 2007 02:04:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
[Kasterisk1*CLI>
<-- SIP read from 64.192.196.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK42d89e6f;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as7793e163
To: <sip:64.192.196.12>;tag=2A07F49B-1E35
Date: Sat, 09 Mar 2002 03:52:44 GMT
Call-ID: 07dbb55552be22af4750756258b79e06@208.75.208.90
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,resource-priority,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 450
v=0
o=CiscoSystemsSIP-GW-UserAgent 820 3931 IN IP4 64.192.196.12
s=SIP Call
c=IN IP4 64.192.196.12
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 15 3
c=IN IP4 64.192.196.12
m=image 0 udptl t38
c=IN IP4 64.192.196.12
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
[Kasterisk1*CLI>
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
--- (14 headers 18 lines) ---
[Kasterisk1*CLI>
Destroying call '07dbb55552be22af4750756258b79e06@208.75.208.90'
[Kasterisk1*CLI>
12 headers, 0 lines
[Kasterisk1*CLI>
Reliably Transmitting (NAT) to 64.192.196.12:5060:
OPTIONS sip:64.192.196.12 SIP/2.0
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK6039c243;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as2e298773
To: <sip:64.192.196.12>
Contact: <sip:Unknown@208.75.208.90>
Call-ID: 2b3136b7539733ec38a5e00f7325d3cc@208.75.208.90
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 09 Nov 2007 02:05:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
[Kasterisk1*CLI>
<-- SIP read from 64.192.196.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK6039c243;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as2e298773
To: <sip:64.192.196.12>;tag=2A08DEF1-23CB
Date: Sat, 09 Mar 2002 03:53:44 GMT
Call-ID: 2b3136b7539733ec38a5e00f7325d3cc@208.75.208.90
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,resource-priority,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 451
v=0
o=CiscoSystemsSIP-GW-UserAgent 4842 5463 IN IP4 64.192.196.12
s=SIP Call
c=IN IP4 64.192.196.12
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 15 3
c=IN IP4 64.192.196.12
m=image 0 udptl t38
c=IN IP4 64.192.196.12
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
--- (14 headers 18 lines) ---
[Kasterisk1*CLI>
Destroying call '2b3136b7539733ec38a5e00f7325d3cc@208.75.208.90'
[Kasterisk1*CLI>
12 headers, 0 lines
Reliably Transmitting (NAT) to 64.192.196.12:5060:
OPTIONS sip:64.192.196.12 SIP/2.0
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK0cc7835e;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as66884b92
To: <sip:64.192.196.12>
Contact: <sip:Unknown@208.75.208.90>
Call-ID: 17f6441a4c65ce7023427319622a6248@208.75.208.90
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 09 Nov 2007 02:05:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
[Kasterisk1*CLI>
<-- SIP read from 64.192.196.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK0cc7835e;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as66884b92
To: <sip:64.192.196.12>;tag=2A08DF2D-16A2
Date: Sat, 09 Mar 2002 03:53:44 GMT
Call-ID: 17f6441a4c65ce7023427319622a6248@208.75.208.90
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,resource-priority,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 451
v=0
o=CiscoSystemsSIP-GW-UserAgent 7777 9779 IN IP4 64.192.196.12
s=SIP Call
c=IN IP4 64.192.196.12
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 15 3
c=IN IP4 64.192.196.12
m=image 0 udptl t38
c=IN IP4 64.192.196.12
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
--- (14 headers 18 lines) ---
[Kasterisk1*CLI>
Destroying call '17f6441a4c65ce7023427319622a6248@208.75.208.90'
[Kasterisk1*CLI>
12 headers, 0 lines
Reliably Transmitting (NAT) to 64.192.196.12:5060:
OPTIONS sip:64.192.196.12 SIP/2.0
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK079cc1e6;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as5fd65e9e
To: <sip:64.192.196.12>
Contact: <sip:Unknown@208.75.208.90>
Call-ID: 5e46fa8a0693d2c724cd3db748390744@208.75.208.90
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 09 Nov 2007 02:06:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
[Kasterisk1*CLI>
<-- SIP read from 64.192.196.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK079cc1e6;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as5fd65e9e
To: <sip:64.192.196.12>;tag=2A09C984-103C
Date: Sat, 09 Mar 2002 03:54:44 GMT
Call-ID: 5e46fa8a0693d2c724cd3db748390744@208.75.208.90
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,resource-priority,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 450
v=0
o=CiscoSystemsSIP-GW-UserAgent 324 7525 IN IP4 64.192.196.12
s=SIP Call
c=IN IP4 64.192.196.12
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 15 3
c=IN IP4 64.192.196.12
m=image 0 udptl t38
c=IN IP4 64.192.196.12
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
[Kasterisk1*CLI>
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
--- (14 headers 18 lines) ---
Destroying call '5e46fa8a0693d2c724cd3db748390744@208.75.208.90'
[Kasterisk1*CLI>
12 headers, 0 lines
Reliably Transmitting (NAT) to 64.192.196.12:5060:
OPTIONS sip:64.192.196.12 SIP/2.0
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK0109b0b0;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as47fa17da
To: <sip:64.192.196.12>
Contact: <sip:Unknown@208.75.208.90>
Call-ID: 442070d0002683c52a102519024aae08@208.75.208.90
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 09 Nov 2007 02:06:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
[Kasterisk1*CLI>
<-- SIP read from 64.192.196.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK0109b0b0;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as47fa17da
To: <sip:64.192.196.12>;tag=2A09C9BC-79
Date: Sat, 09 Mar 2002 03:54:44 GMT
Call-ID: 442070d0002683c52a102519024aae08@208.75.208.90
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,resource-priority,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 451
v=0
o=CiscoSystemsSIP-GW-UserAgent 7453 4789 IN IP4 64.192.196.12
s=SIP Call
c=IN IP4 64.192.196.12
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 15 3
c=IN IP4 64.192.196.12
m=image 0 udptl t38
c=IN IP4 64.192.196.12
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a
[Kasterisk1*CLI>
=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
--- (14 headers 18 lines) ---
[Kasterisk1*CLI>
Destroying call '442070d0002683c52a102519024aae08@208.75.208.90'
[Kasterisk1*CLI>
12 headers, 0 lines
[Kasterisk1*CLI>
Reliably Transmitting (NAT) to 64.192.196.12:5060:
OPTIONS sip:64.192.196.12 SIP/2.0
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK1f67ac39;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as3f8772b0
To: <sip:64.192.196.12>
Contact: <sip:Unknown@208.75.208.90>
Call-ID: 676eab12426acf8252e9ad87090c694c@208.75.208.90
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 09 Nov 2007 02:07:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
[Kasterisk1*CLI>
<-- SIP read from 64.192.196.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK1f67ac39;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as3f8772b0
To: <sip:64.192.196.12>;tag=2A0AB412-12F8
Date: Sat, 09 Mar 2002 03:55:44 GMT
Call-ID: 676eab12426acf8252e9ad87090c694c@208.75.208.90
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,resource-priority,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 450
v=0
o=CiscoSystemsSIP-GW-UserAgent 2114 440 IN IP4 64.192.196.12
s=SIP Call
c=IN IP4 64.192.196.12
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 15 3
c=IN IP4 64.192.196.12
m=image 0 udptl t38
c=IN IP4 64.192.196.12
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
[Kasterisk1*CLI>
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
--- (14 headers 18 lines) ---
[Kasterisk1*CLI>
Destroying call '676eab12426acf8252e9ad87090c694c@208.75.208.90'
[Kasterisk1*CLI>
12 headers, 0 lines
[Kasterisk1*CLI>
Reliably Transmitting (NAT) to 64.192.196.12:5060:
OPTIONS sip:64.192.196.12 SIP/2.0
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK7104591a;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as27faa225
To: <sip:64.192.196.12>
Contact: <sip:Unknown@208.75.208.90>
Call-ID: 6ddd4e1c2b511b3c29bb66fa72e78b7b@208.75.208.90
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 09 Nov 2007 02:07:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
[Kasterisk1*CLI>
<-- SIP read from 64.192.196.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK7104591a;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as27faa225
To: <sip:64.192.196.12>;tag=2A0AB44A-1A3B
Date: Sat, 09 Mar 2002 03:55:44 GMT
Call-ID: 6ddd4e1c2b511b3c29bb66fa72e78b7b@208.75.208.90
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,resource-priority,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 451
v=0
o=CiscoSystemsSIP-GW-UserAgent 7721 8015 IN IP4 64.192.196.12
s=SIP Call
c=IN IP4 64.192.196.12
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 15 3
c=IN IP4 64.192.196.12
m=image 0 udptl t38
c=IN IP4 64.192.196.12
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
--- (14 headers 18 lines) ---
[Kasterisk1*CLI>
Destroying call '6ddd4e1c2b511b3c29bb66fa72e78b7b@208.75.208.90'
[Kasterisk1*CLI>
12 headers, 0 lines
Reliably Transmitting (NAT) to 64.192.196.12:5060:
OPTIONS sip:64.192.196.12 SIP/2.0
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK50801fa2;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as0de10b33
To: <sip:64.192.196.12>
Contact: <sip:Unknown@208.75.208.90>
Call-ID: 6c11ca876bd68f7a667ac8c21b436081@208.75.208.90
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 09 Nov 2007 02:08:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
[Kasterisk1*CLI>
<-- SIP read from 64.192.196.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK50801fa2;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as0de10b33
To: <sip:64.192.196.12>;tag=2A0B9EA0-1EAF
Date: Sat, 09 Mar 2002 03:56:44 GMT
Call-ID: 6c11ca876bd68f7a667ac8c21b436081@208.75.208.90
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,resource-priority,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 451
v=0
o=CiscoSystemsSIP-GW-UserAgent 9284 2127 IN IP4 64.192.196.12
s=SIP Call
c=IN IP4 64.192.196.12
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 15 3
c=IN IP4 64.192.196.12
m=image 0 udptl t38
c=IN IP4 64.192.196.12
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
--- (14 headers 18 lines) ---
[Kasterisk1*CLI>
Destroying call '6c11ca876bd68f7a667ac8c21b436081@208.75.208.90'
[Kasterisk1*CLI>
12 headers, 0 lines
Reliably Transmitting (NAT) to 64.192.196.12:5060:
OPTIONS sip:64.192.196.12 SIP/2.0
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK38a06f04;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as115ce44d
To: <sip:64.192.196.12>
Contact: <sip:Unknown@208.75.208.90>
Call-ID: 6b83d1eb307af0e56b775f7224c60a68@208.75.208.90
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 09 Nov 2007 02:08:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
[Kasterisk1*CLI>
<-- SIP read from 64.192.196.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK38a06f04;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as115ce44d
To: <sip:64.192.196.12>;tag=2A0B9ED9-2082
Date: Sat, 09 Mar 2002 03:56:44 GMT
Call-ID: 6b83d1eb307af0e56b775f7224c60a68@208.75.208.90
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,resource-priority,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 451
v=0
o=CiscoSystemsSIP-GW-UserAgent 9163 3868 IN IP4 64.192.196.12
s=SIP Call
c=IN IP4 64.192.196.12
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 15 3
c=IN IP4 64.192.196.12
m=image 0 udptl t38
c=IN IP4 64.192.196.12
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
--- (14 headers 18 lines) ---
[Kasterisk1*CLI>
Destroying call '6b83d1eb307af0e56b775f7224c60a68@208.75.208.90'
[Kasterisk1*CLI> riority,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 451
v=0
o=CiscoSystemsSIP-GW-UserAgent 4322 5260 IN IP4 64.192.196.12
s=SIP Call
c=IN IP4 64.192.196.12
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 15 3
c=IN IP4 64.192.196.12
m=image 0 udptl t38
c=IN IP4 64.192.196.12
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
[Kasterisk1*CLI>
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
--- (14 headers 18 lines) ---
[Kasterisk1*CLI>
Destroying call '627d5f6b255de2b56b87efa6153d4e5b@208.75.208.90'
[Kasterisk1*CLI>
12 headers, 0 lines
Reliably Transmitting (NAT) to 64.192.196.12:5060:
OPTIONS sip:64.192.196.12 SIP/2.0
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK1118e583;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as23432d27
To: <sip:64.192.196.12>
Contact: <sip:Unknown@208.75.208.90>
Call-ID: 2a492cae314700f938c11a286b21fce7@208.75.208.90
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 09 Nov 2007 02:09:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
[Kasterisk1*CLI>
<-- SIP read from 64.192.196.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK1118e583;rport
From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as23432d27
To: <sip:64.192.196.12>;tag=2A0C8967-1302
Date: Sat, 09 Mar 2002 03:57:44 GMT
Call-ID: 2a492cae314700f938c11a286b21fce7@208.75.208.90
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,resource-priority,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 449
v=0
o=CiscoSystemsSIP-GW-UserAgent 7412 77 IN IP4 64.192.196.12
s=SIP Call
c=IN IP4 64.192.196.12
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 15 3
c=IN IP4 64.192.196.12
m=image 0 udptl t38
c=IN IP4 64.192.196.12
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a
[Kasterisk1*CLI>
=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
--- (14 headers 18 lines) ---
Destroying call '2a492cae314700f938c11a286b21fce7@208.75.208.90'
[Kasterisk1*CLI>