Asterisk 1.4.42, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
[Jul 14 11:13:22] Set to realtime thread
[0;37;40m[Jul 14 11:13:22]   == Parsing '/etc/asterisk/extconfig.conf': [Jul 14 11:13:22] Found
[0m[Jul 14 11:13:22] Connected to Asterisk 1.4.42 currently running on a02-alpha (pid = 6448)
a02-alpha*CLI> 
[0KVerbosity is at least 2

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:23]   == Spawn extension (fromtrunk_g1, 8452900000, 2) exited non-zero on 'DAHDI/4-1'

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:23] Audio is at 192.168.2.10 port 12728

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:23] Adding codec 0x8 (alaw) to SDP

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:23] Reliably Transmitting (no NAT) to 192.168.2.90:5060:
INVITE sip:8452900000@192.168.2.90 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK3fee89e1;rport
From: "89873213690" <sip:alpha-centre@192.168.2.10>;tag=as51d1d387
To: <sip:8452900000@192.168.2.90>
Contact: <sip:alpha-centre@192.168.2.10>
Call-ID: 5b9e3dba75955e6e321fa57a266f1b0c@192.168.2.10
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 14 Jul 2013 07:13:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 155

v=0
o=root 6448 6448 IN IP4 192.168.2.10
s=session
c=IN IP4 192.168.2.10
t=0 0
m=audio 12728 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

---

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:23] 
<--- SIP read from 192.168.2.90:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK3fee89e1;received=192.168.2.10;rport=5060
From: "89873213690" <sip:alpha-centre@192.168.2.10>;tag=as51d1d387
To: <sip:8452900000@192.168.2.90>
Call-ID: 5b9e3dba75955e6e321fa57a266f1b0c@192.168.2.10
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:8452900000@192.168.2.90:5060>
Content-Length: 0


<------------->

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:23] --- (11 headers 0 lines) ---

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:23] 
<--- SIP read from 192.168.2.90:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK3fee89e1;received=192.168.2.10;rport=5060
From: "89873213690" <sip:alpha-centre@192.168.2.10>;tag=as51d1d387
To: <sip:8452900000@192.168.2.90>;tag=as08b10079
Call-ID: 5b9e3dba75955e6e321fa57a266f1b0c@192.168.2.10
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:8452900000@192.168.2.90:5060>
Content-Type: application/sdp
Content-Length: 206

v=0
o=root 417939528 417939528 IN IP4 192.168.2.90
s=Asterisk PBX 1.8.20.0
c=IN IP4 192.168.2.90
t=0 0
m=audio 15468 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:23] --- (12 headers 10 lines) ---

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:23] Found RTP audio format 8

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:23] Found audio description format PCMA for ID 8

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:23] Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:23] Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:23] Peer audio RTP is at port 192.168.2.90:15468

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:23] list_route: hop: <sip:8452900000@192.168.2.90:5060>

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:23] set_destination: Parsing <sip:8452900000@192.168.2.90:5060> for address/port to send to

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:23] set_destination: set destination to 192.168.2.90, port 5060

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:23] Transmitting (no NAT) to 192.168.2.90:5060:
ACK sip:8452900000@192.168.2.90:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK1724f893;rport
From: "89873213690" <sip:alpha-centre@192.168.2.10>;tag=as51d1d387
To: <sip:8452900000@192.168.2.90>;tag=as08b10079
Contact: <sip:alpha-centre@192.168.2.10>
Call-ID: 5b9e3dba75955e6e321fa57a266f1b0c@192.168.2.10
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:27]   == Spawn extension (fromtrunk_g1, 8452900000, 2) exited non-zero on 'DAHDI/3-1'

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:28] Really destroying SIP dialog '48c523d05253e6073d101f9c5cf20e78@192.168.2.90:5060' Method: OPTIONS

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:29] Audio is at 192.168.2.10 port 17544

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:29] Adding codec 0x8 (alaw) to SDP

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:29] Reliably Transmitting (no NAT) to 192.168.2.90:5060:
INVITE sip:8452900000@192.168.2.90 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK75d14a7c;rport
From: "89616447484" <sip:alpha-centre@192.168.2.10>;tag=as4c1247a6
To: <sip:8452900000@192.168.2.90>
Contact: <sip:alpha-centre@192.168.2.10>
Call-ID: 4298227e1c2fcc8b0b287115245267e5@192.168.2.10
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 14 Jul 2013 07:13:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 155

v=0
o=root 6448 6448 IN IP4 192.168.2.10
s=session
c=IN IP4 192.168.2.10
t=0 0
m=audio 17544 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

---

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:29] 
<--- SIP read from 192.168.2.90:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK75d14a7c;received=192.168.2.10;rport=5060
From: "89616447484" <sip:alpha-centre@192.168.2.10>;tag=as4c1247a6
To: <sip:8452900000@192.168.2.90>
Call-ID: 4298227e1c2fcc8b0b287115245267e5@192.168.2.10
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:8452900000@192.168.2.90:5060>
Content-Length: 0


<------------->

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:29] --- (11 headers 0 lines) ---

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:29] 
<--- SIP read from 192.168.2.90:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK75d14a7c;received=192.168.2.10;rport=5060
From: "89616447484" <sip:alpha-centre@192.168.2.10>;tag=as4c1247a6
To: <sip:8452900000@192.168.2.90>;tag=as40f8ec10
Call-ID: 4298227e1c2fcc8b0b287115245267e5@192.168.2.10
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:8452900000@192.168.2.90:5060>
Content-Type: application/sdp
Content-Length: 206

v=0
o=root 754688353 754688353 IN IP4 192.168.2.90
s=Asterisk PBX 1.8.20.0
c=IN IP4 192.168.2.90
t=0 0
m=audio 12928 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:29] --- (12 headers 10 lines) ---

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:29] Found RTP audio format 8

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:29] Found audio description format PCMA for ID 8

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:29] Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:29] Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:29] Peer audio RTP is at port 192.168.2.90:12928

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:29] list_route: hop: <sip:8452900000@192.168.2.90:5060>

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:29] set_destination: Parsing <sip:8452900000@192.168.2.90:5060> for address/port to send to

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:29] set_destination: set destination to 192.168.2.90, port 5060

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:29] Transmitting (no NAT) to 192.168.2.90:5060:
ACK sip:8452900000@192.168.2.90:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK73c3fff7;rport
From: "89616447484" <sip:alpha-centre@192.168.2.10>;tag=as4c1247a6
To: <sip:8452900000@192.168.2.90>;tag=as40f8ec10
Contact: <sip:alpha-centre@192.168.2.10>
Call-ID: 4298227e1c2fcc8b0b287115245267e5@192.168.2.10
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:30]   == Spawn extension (fromtrunk_g1, 8452900000, 2) exited non-zero on 'DAHDI/1-1'

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:35]   == Spawn extension (to-ttk, 89020413665, 6) exited non-zero on 'SIP/8452392268-00161714'

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:38] Audio is at 192.168.2.10 port 18562

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:38] Adding codec 0x8 (alaw) to SDP

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:38] Reliably Transmitting (no NAT) to 192.168.2.90:5060:
INVITE sip:8452900000@192.168.2.90 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK157c910f;rport
From: "89372542388" <sip:alpha-centre@192.168.2.10>;tag=as0e95b636
To: <sip:8452900000@192.168.2.90>
Contact: <sip:alpha-centre@192.168.2.10>
Call-ID: 5a082ca80d6daaa3128cc4dd5e8af1af@192.168.2.10
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 14 Jul 2013 07:13:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 155

v=0
o=root 6448 6448 IN IP4 192.168.2.10
s=session
c=IN IP4 192.168.2.10
t=0 0
m=audio 18562 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

---

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:38] 
<--- SIP read from 192.168.2.90:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK157c910f;received=192.168.2.10;rport=5060
From: "89372542388" <sip:alpha-centre@192.168.2.10>;tag=as0e95b636
To: <sip:8452900000@192.168.2.90>
Call-ID: 5a082ca80d6daaa3128cc4dd5e8af1af@192.168.2.10
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:8452900000@192.168.2.90:5060>
Content-Length: 0


<------------->

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:38] --- (11 headers 0 lines) ---

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:38] 
<--- SIP read from 192.168.2.90:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK157c910f;received=192.168.2.10;rport=5060
From: "89372542388" <sip:alpha-centre@192.168.2.10>;tag=as0e95b636
To: <sip:8452900000@192.168.2.90>;tag=as43efcb92
Call-ID: 5a082ca80d6daaa3128cc4dd5e8af1af@192.168.2.10
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:8452900000@192.168.2.90:5060>
Content-Type: application/sdp
Content-Length: 208

v=0
o=root 1421345798 1421345798 IN IP4 192.168.2.90
s=Asterisk PBX 1.8.20.0
c=IN IP4 192.168.2.90
t=0 0
m=audio 19836 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:38] --- (12 headers 10 lines) ---

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:38] Found RTP audio format 8

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:38] Found audio description format PCMA for ID 8

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:38] Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:38] Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:38] Peer audio RTP is at port 192.168.2.90:19836

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:38] list_route: hop: <sip:8452900000@192.168.2.90:5060>

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:38] set_destination: Parsing <sip:8452900000@192.168.2.90:5060> for address/port to send to

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:38] set_destination: set destination to 192.168.2.90, port 5060

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:38] Transmitting (no NAT) to 192.168.2.90:5060:
ACK sip:8452900000@192.168.2.90:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK45c32f95;rport
From: "89372542388" <sip:alpha-centre@192.168.2.10>;tag=as0e95b636
To: <sip:8452900000@192.168.2.90>;tag=as43efcb92
Contact: <sip:alpha-centre@192.168.2.10>
Call-ID: 5a082ca80d6daaa3128cc4dd5e8af1af@192.168.2.10
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:39] Audio is at 192.168.2.10 port 12782

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:39] Adding codec 0x8 (alaw) to SDP

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:39] Reliably Transmitting (no NAT) to 192.168.2.90:5060:
INVITE sip:8452900000@192.168.2.90 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK52e5fa2c;rport
From: "89172011440" <sip:alpha-centre@192.168.2.10>;tag=as33a2078b
To: <sip:8452900000@192.168.2.90>
Contact: <sip:alpha-centre@192.168.2.10>
Call-ID: 0e7f74aa1f6d814863ccc98c132f17ed@192.168.2.10
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 14 Jul 2013 07:13:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 155

v=0
o=root 6448 6448 IN IP4 192.168.2.10
s=session
c=IN IP4 192.168.2.10
t=0 0
m=audio 12782 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

---

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:39] 
<--- SIP read from 192.168.2.90:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK52e5fa2c;received=192.168.2.10;rport=5060
From: "89172011440" <sip:alpha-centre@192.168.2.10>;tag=as33a2078b
To: <sip:8452900000@192.168.2.90>
Call-ID: 0e7f74aa1f6d814863ccc98c132f17ed@192.168.2.10
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:8452900000@192.168.2.90:5060>
Content-Length: 0


<------------->

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:39] --- (11 headers 0 lines) ---

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:39] 
<--- SIP read from 192.168.2.90:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK52e5fa2c;received=192.168.2.10;rport=5060
From: "89172011440" <sip:alpha-centre@192.168.2.10>;tag=as33a2078b
To: <sip:8452900000@192.168.2.90>;tag=as2b1bf27d
Call-ID: 0e7f74aa1f6d814863ccc98c132f17ed@192.168.2.10
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:8452900000@192.168.2.90:5060>
Content-Type: application/sdp
Content-Length: 208

v=0
o=root 1056253780 1056253780 IN IP4 192.168.2.90
s=Asterisk PBX 1.8.20.0
c=IN IP4 192.168.2.90
t=0 0
m=audio 10086 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:39] --- (12 headers 10 lines) ---

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:39] Found RTP audio format 8

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:39] Found audio description format PCMA for ID 8

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:39] Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:39] Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:39] Peer audio RTP is at port 192.168.2.90:10086

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:39] list_route: hop: <sip:8452900000@192.168.2.90:5060>

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:39] set_destination: Parsing <sip:8452900000@192.168.2.90:5060> for address/port to send to

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:39] set_destination: set destination to 192.168.2.90, port 5060

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:39] Transmitting (no NAT) to 192.168.2.90:5060:
ACK sip:8452900000@192.168.2.90:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK78a58464;rport
From: "89172011440" <sip:alpha-centre@192.168.2.10>;tag=as33a2078b
To: <sip:8452900000@192.168.2.90>;tag=as2b1bf27d
Contact: <sip:alpha-centre@192.168.2.10>
Call-ID: 0e7f74aa1f6d814863ccc98c132f17ed@192.168.2.10
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:44] 
<--- SIP read from 192.168.2.90:5060 --->
BYE sip:alpha-centre@192.168.2.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK1ba32990;rport
Max-Forwards: 70
From: <sip:8452900000@192.168.2.90>;tag=as08b10079
To: "89873213690" <sip:alpha-centre@192.168.2.10>;tag=as51d1d387
Call-ID: 5b9e3dba75955e6e321fa57a266f1b0c@192.168.2.10
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.8.20.0
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 0
Content-Length: 0


<------------->

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:44] --- (11 headers 0 lines) ---

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:44] Sending to 192.168.2.90 : 5060 (no NAT)

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:44] 
<--- Transmitting (no NAT) to 192.168.2.90:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK1ba32990;received=192.168.2.90;rport=5060
From: <sip:8452900000@192.168.2.90>;tag=as08b10079
To: "89873213690" <sip:alpha-centre@192.168.2.10>;tag=as51d1d387
Call-ID: 5b9e3dba75955e6e321fa57a266f1b0c@192.168.2.10
CSeq: 102 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------>

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:44]   == Spawn extension (to_new, 8452900000, 4) exited non-zero on 'DAHDI/5-1'

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:44] Really destroying SIP dialog '5b9e3dba75955e6e321fa57a266f1b0c@192.168.2.10' Method: BYE

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:45] 
<--- SIP read from 192.168.2.90:5060 --->
BYE sip:alpha-centre@192.168.2.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK1eeec0d6;rport
Max-Forwards: 70
From: <sip:8452900000@192.168.2.90>;tag=as40f8ec10
To: "89616447484" <sip:alpha-centre@192.168.2.10>;tag=as4c1247a6
Call-ID: 4298227e1c2fcc8b0b287115245267e5@192.168.2.10
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.8.20.0
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 0
Content-Length: 0


<------------->

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:45] --- (11 headers 0 lines) ---

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:45] Sending to 192.168.2.90 : 5060 (no NAT)

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:45] 
<--- Transmitting (no NAT) to 192.168.2.90:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK1eeec0d6;received=192.168.2.90;rport=5060
From: <sip:8452900000@192.168.2.90>;tag=as40f8ec10
To: "89616447484" <sip:alpha-centre@192.168.2.10>;tag=as4c1247a6
Call-ID: 4298227e1c2fcc8b0b287115245267e5@192.168.2.10
CSeq: 102 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------>

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:45]   == Spawn extension (to_new, 8452900000, 4) exited non-zero on 'DAHDI/6-1'

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:45] Really destroying SIP dialog '4298227e1c2fcc8b0b287115245267e5@192.168.2.10' Method: BYE

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:48]   == Spawn extension (to-ttk, 89518842377, 5) exited non-zero on 'SIP/8452392268-0016170f'

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:49] Audio is at 192.168.2.10 port 13416

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:49] Adding codec 0x8 (alaw) to SDP

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:49] Reliably Transmitting (no NAT) to 192.168.2.90:5060:
INVITE sip:8452900000@192.168.2.90 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK2b9d69c2;rport
From: "89085570124" <sip:alpha-centre@192.168.2.10>;tag=as4a01208c
To: <sip:8452900000@192.168.2.90>
Contact: <sip:alpha-centre@192.168.2.10>
Call-ID: 6ae8a8c42ab395e75280cbe56869f843@192.168.2.10
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 14 Jul 2013 07:13:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 155

v=0
o=root 6448 6448 IN IP4 192.168.2.10
s=session
c=IN IP4 192.168.2.10
t=0 0
m=audio 13416 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

---

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:49] 
<--- SIP read from 192.168.2.90:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK2b9d69c2;received=192.168.2.10;rport=5060
From: "89085570124" <sip:alpha-centre@192.168.2.10>;tag=as4a01208c
To: <sip:8452900000@192.168.2.90>
Call-ID: 6ae8a8c42ab395e75280cbe56869f843@192.168.2.10
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:8452900000@192.168.2.90:5060>
Content-Length: 0


<------------->

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:49] --- (11 headers 0 lines) ---

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:49] 
<--- SIP read from 192.168.2.90:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK2b9d69c2;received=192.168.2.10;rport=5060
From: "89085570124" <sip:alpha-centre@192.168.2.10>;tag=as4a01208c
To: <sip:8452900000@192.168.2.90>;tag=as69b81870
Call-ID: 6ae8a8c42ab395e75280cbe56869f843@192.168.2.10
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:8452900000@192.168.2.90:5060>
Content-Type: application/sdp
Content-Length: 208

v=0
o=root 1628471683 1628471683 IN IP4 192.168.2.90
s=Asterisk PBX 1.8.20.0
c=IN IP4 192.168.2.90
t=0 0
m=audio 10492 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:49] --- (12 headers 10 lines) ---

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:49] Found RTP audio format 8

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:49] Found audio description format PCMA for ID 8

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:49] Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:49] Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:49] Peer audio RTP is at port 192.168.2.90:10492

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:49] list_route: hop: <sip:8452900000@192.168.2.90:5060>

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:49] set_destination: Parsing <sip:8452900000@192.168.2.90:5060> for address/port to send to

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:49] set_destination: set destination to 192.168.2.90, port 5060

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:49] Transmitting (no NAT) to 192.168.2.90:5060:
ACK sip:8452900000@192.168.2.90:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK2c13b094;rport
From: "89085570124" <sip:alpha-centre@192.168.2.10>;tag=as4a01208c
To: <sip:8452900000@192.168.2.90>;tag=as69b81870
Contact: <sip:alpha-centre@192.168.2.10>
Call-ID: 6ae8a8c42ab395e75280cbe56869f843@192.168.2.10
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:52]   == Spawn extension (fromtrunk_g1, 8452900000, 2) exited non-zero on 'DAHDI/26-1'

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:54] Audio is at 192.168.2.10 port 15676

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:54] Adding codec 0x8 (alaw) to SDP

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:54] Reliably Transmitting (no NAT) to 192.168.2.90:5060:
INVITE sip:8452900000@192.168.2.90 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK597e55e5;rport
From: "89276220039" <sip:alpha-centre@192.168.2.10>;tag=as7e8abad0
To: <sip:8452900000@192.168.2.90>
Contact: <sip:alpha-centre@192.168.2.10>
Call-ID: 447ff969744d92fb4653955866a164ed@192.168.2.10
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 14 Jul 2013 07:13:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 155

v=0
o=root 6448 6448 IN IP4 192.168.2.10
s=session
c=IN IP4 192.168.2.10
t=0 0
m=audio 15676 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

---

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:54] 
<--- SIP read from 192.168.2.90:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK597e55e5;received=192.168.2.10;rport=5060
From: "89276220039" <sip:alpha-centre@192.168.2.10>;tag=as7e8abad0
To: <sip:8452900000@192.168.2.90>
Call-ID: 447ff969744d92fb4653955866a164ed@192.168.2.10
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:8452900000@192.168.2.90:5060>
Content-Length: 0


<------------->

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:54] --- (11 headers 0 lines) ---

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:54] 
<--- SIP read from 192.168.2.90:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK597e55e5;received=192.168.2.10;rport=5060
From: "89276220039" <sip:alpha-centre@192.168.2.10>;tag=as7e8abad0
To: <sip:8452900000@192.168.2.90>;tag=as2ff8c8ae
Call-ID: 447ff969744d92fb4653955866a164ed@192.168.2.10
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:8452900000@192.168.2.90:5060>
Content-Type: application/sdp
Content-Length: 208

v=0
o=root 1881275455 1881275455 IN IP4 192.168.2.90
s=Asterisk PBX 1.8.20.0
c=IN IP4 192.168.2.90
t=0 0
m=audio 11470 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:54] --- (12 headers 10 lines) ---

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:54] Found RTP audio format 8
[2013-07-14 11:13:54] Found audio description format PCMA for ID 8

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:54] Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:54] Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:54] Peer audio RTP is at port 192.168.2.90:11470

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:54] list_route: hop: <sip:8452900000@192.168.2.90:5060>

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:54] set_destination: Parsing <sip:8452900000@192.168.2.90:5060> for address/port to send to

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:54] set_destination: set destination to 192.168.2.90, port 5060

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:54] Transmitting (no NAT) to 192.168.2.90:5060:
ACK sip:8452900000@192.168.2.90:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK64ebd683;rport
From: "89276220039" <sip:alpha-centre@192.168.2.10>;tag=as7e8abad0
To: <sip:8452900000@192.168.2.90>;tag=as2ff8c8ae
Contact: <sip:alpha-centre@192.168.2.10>
Call-ID: 447ff969744d92fb4653955866a164ed@192.168.2.10
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:56] 
<--- SIP read from 192.168.2.90:5060 --->
OPTIONS sip:192.168.2.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK74729595
Max-Forwards: 70
From: "asterisk" <sip:alpha-centre@192.168.2.90>;tag=as5fbd7144
To: <sip:192.168.2.10>
Contact: <sip:alpha-centre@192.168.2.90:5060>
Call-ID: 0de809e36269eaf222bbe13243c4a70a@192.168.2.90:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.20.0
Date: Sun, 14 Jul 2013 07:12:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------->

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:56] --- (13 headers 0 lines) ---

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:56] Looking for s in default (domain 192.168.2.10)

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:56] 
<--- Transmitting (no NAT) to 192.168.2.90:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK74729595;received=192.168.2.90
From: "asterisk" <sip:alpha-centre@192.168.2.90>;tag=as5fbd7144
To: <sip:192.168.2.10>;tag=as31c1b58a
Call-ID: 0de809e36269eaf222bbe13243c4a70a@192.168.2.90:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:192.168.2.10>
Accept: application/sdp
Content-Length: 0


<------------>

[Ka02-alpha*CLI> 
[0K[2013-07-14 11:13:56] Scheduling destruction of SIP dialog '0de809e36269eaf222bbe13243c4a70a@192.168.2.90:5060' in 32000 ms (Method: OPTIONS)

[Ka02-alpha*CLI> 
Disconnected from Asterisk server
[Jul 14 11:13:59] Executing last minute cleanups
[Jul 14 11:13:59] Asterisk cleanly ending (0).
[0m