Asterisk 1.8.20.0, Copyright (C) 1999 - 2012 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.
=========================================================================
Connected to Asterisk 1.8.20.0 currently running on a01-centre (pid = 30039)
a01-centre*CLI> 
[0KVerbosity is at least 3
Core debug is at least 3

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.38:5651 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK653e1029
From: "89873213690"<sip:alpha-centre@192.168.2.90>;tag=as1be3e29c
Call-ID: 581e47ea28cc941d7044d089575b49f5@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2085@192.168.0.38:5651>;tag=memutrwl
Content-Type: application/sdp
Content-Length: 198
Contact: <sip:2085@192.168.0.38:5651>

v=0
o=- 131597603 131597603 IN IP4 192.168.0.38
s=Ozeki VoIP SIP SDK
c=IN IP4 192.168.0.38
t=0 0
m=audio 5729 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
--- (9 headers 9 lines) ---

[Ka01-centre*CLI> 
[0KFound RTP audio format 3
Found RTP audio format 101
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - 0x2 (gsm), peer - audio=0x2 (gsm)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x2 (gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.38:5729

[Ka01-centre*CLI> 
[0Klist_route: hop: <sip:2085@192.168.0.38:5651>

[Ka01-centre*CLI> 
[0Kset_destination: Parsing <sip:2085@192.168.0.38:5651> for address/port to send to
set_destination: set destination to 192.168.0.38:5651
Transmitting (no NAT) to 192.168.0.38:5651:
ACK sip:2085@192.168.0.38:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK56abeaa0
Max-Forwards: 70
From: "89873213690" <sip:alpha-centre@192.168.2.90>;tag=as1be3e29c
To: <sip:2085@192.168.0.38:5651>;tag=memutrwl
Contact: <sip:alpha-centre@192.168.2.90:5060>
Call-ID: 581e47ea28cc941d7044d089575b49f5@192.168.2.90:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.20.0
Content-Length: 0


---

[Ka01-centre*CLI> 
[0K    -- SIP/2085-0000f937 answered SIP/alpha-centre-0000f935

[Ka01-centre*CLI> 
[0K    -- Stopped music on hold on SIP/alpha-centre-0000f935

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:10.1.0.2:1028 --->
BYE sip:8452753083@192.168.2.90:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.2:1028;branch=z9hG4bK343ae829
Max-Forwards: 70
From: "89658820359" <sip:astr-centre@10.1.0.2:1028>;tag=as61232dad
To: <sip:8452753083@192.168.2.90>;tag=as24f081c4
Call-ID: 31a1fbf25b4024fd7f195aa67cfbea09@192.168.5.33:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.8.20.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

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

[Ka01-centre*CLI> 
[0K--- (11 headers 0 lines) ---
Sending to 10.1.0.2:1028 (no NAT)

[Ka01-centre*CLI> 
[0KScheduling destruction of SIP dialog '31a1fbf25b4024fd7f195aa67cfbea09@192.168.5.33:5060' in 6400 ms (Method: BYE)

<--- Transmitting (no NAT) to 10.1.0.2:1028 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.0.2:1028;branch=z9hG4bK343ae829;received=10.1.0.2
From: "89658820359" <sip:astr-centre@10.1.0.2:1028>;tag=as61232dad
To: <sip:8452753083@192.168.2.90>;tag=as24f081c4
Call-ID: 31a1fbf25b4024fd7f195aa67cfbea09@192.168.5.33:5060
CSeq: 103 BYE
Server: Asterisk PBX 1.8.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


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

[Ka01-centre*CLI> 
[0KScheduling destruction of SIP dialog '18a73cab689c8e1110585ffc53d7b30c@192.168.2.90:5060' in 6400 ms (Method: INVITE)

[Ka01-centre*CLI> 
[0Kset_destination: Parsing <sip:8452440000@192.168.2.60:5060> for address/port to send to

[Ka01-centre*CLI> 
[0Kset_destination: set destination to 192.168.2.60:5060

[Ka01-centre*CLI> 
[0KReliably Transmitting (no NAT) to 192.168.2.60:5060:
BYE sip:8452440000@192.168.2.60:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK064c47cf
Max-Forwards: 70
From: "89658820359" <sip:epsilon@192.168.2.90>;tag=as445e697e
To: <sip:8452440000@192.168.2.60>;tag=as03557b98
Call-ID: 18a73cab689c8e1110585ffc53d7b30c@192.168.2.90:5060
CSeq: 104 BYE
User-Agent: Asterisk PBX 1.8.20.0
Authorization: Digest username="epsilon", realm="asterisk", algorithm=MD5, uri="sip:8452440000@192.168.2.60:5060", nonce="44840bc0", response="7448e12d021290f6e7a77245bb84d095"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

[Ka01-centre*CLI> 
[0K  == Spawn extension (in_dial, 8452753083, 29) exited non-zero on 'SIP/astr-centre-0000f92f'

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.2.60:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK064c47cf;received=192.168.2.90;rport=5060
From: "89658820359" <sip:epsilon@192.168.2.90>;tag=as445e697e
To: <sip:8452440000@192.168.2.60>;tag=as03557b98
Call-ID: 18a73cab689c8e1110585ffc53d7b30c@192.168.2.90:5060
CSeq: 104 BYE
Server: Asterisk PBX 1.8.10.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

[Ka01-centre*CLI> 
[0KReally destroying SIP dialog '18a73cab689c8e1110585ffc53d7b30c@192.168.2.90:5060' Method: INVITE

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:10.1.0.2:1028 --->
OPTIONS sip:192.168.2.90 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.2:1028;branch=z9hG4bK4523fcb9
Max-Forwards: 70
From: "asterisk" <sip:astr-centre@10.1.0.2:1028>;tag=as4fc788d4
To: <sip:192.168.2.90>
Contact: <sip:astr-centre@10.1.0.2:1028>
Call-ID: 787c44ba472d0db5697051c372edee18@192.168.5.33:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.20.0
Date: Sun, 14 Jul 2013 07:11:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Looking for s in default (domain 192.168.2.90)

[Ka01-centre*CLI> 
[0K
<--- Transmitting (NAT) to 10.1.0.2:1028 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.0.2:1028;branch=z9hG4bK4523fcb9;received=10.1.0.2;rport=1028
From: "asterisk" <sip:astr-centre@10.1.0.2:1028>;tag=as4fc788d4
To: <sip:192.168.2.90>;tag=as7830642f
Call-ID: 787c44ba472d0db5697051c372edee18@192.168.5.33:5060
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.8.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:192.168.2.90:5060>
Accept: application/sdp
Content-Length: 0


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

[Ka01-centre*CLI> 
[0KScheduling destruction of SIP dialog '787c44ba472d0db5697051c372edee18@192.168.5.33:5060' in 32000 ms (Method: OPTIONS)

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.2.10:5060 --->
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
<------------->
--- (14 headers 9 lines) ---

[Ka01-centre*CLI> 
[0KSending to 192.168.2.10:5060 (NAT)
Using INVITE request as basis request - 4298227e1c2fcc8b0b287115245267e5@192.168.2.10

[Ka01-centre*CLI> 
[0KFound peer 'alpha-centre' for 'alpha-centre' from 192.168.2.10:5060

[Ka01-centre*CLI> 
[0K  == Using SIP RTP CoS mark 5

[Ka01-centre*CLI> 
[0KFound RTP audio format 8
Found audio description format PCMA for ID 8
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.2.10:17544
Looking for 8452900000 in in_dial (domain 192.168.2.90)

[Ka01-centre*CLI> 
[0Klist_route: hop: <sip:alpha-centre@192.168.2.10>

<--- Transmitting (no NAT) to 192.168.2.10:5060 --->
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


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

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:1] [1;36mNoOp[0m("[1;35mSIP/alpha-centre-0000f938[0m", "[1;35m<<<=== Входящий 89616447484 a01-centre-1373785895.63800 <<<===[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:2] [1;36mSystem[0m("[1;35mSIP/alpha-centre-0000f938[0m", "[1;35m/bin/echo "`date "+%F_%H-%M-%S"` "89616447484" <alpha-centre>" >> /tmp/calls-N.log[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:3] [1;36mGotoIf[0m("[1;35mSIP/alpha-centre-0000f938[0m", "[1;35m0?To_Old[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:4] [1;36mGotoIf[0m("[1;35mSIP/alpha-centre-0000f938[0m", "[1;35m0?blacklist[0m") in new stack
    -- Executing [8452900000@in_dial:5] [1;36mSet[0m("[1;35mSIP/alpha-centre-0000f938[0m", "[1;35mOSTATE=0[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:6] [1;36mAGI[0m("[1;35mSIP/alpha-centre-0000f938[0m", "[1;35mallo/get_order_state.php,89616447484[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Launched AGI Script /usr/share/asterisk/agi-bin/allo/get_order_state.php

[Ka01-centre*CLI> 
[0K[Jul 14 11:11:35] [0;31mERROR[0m[24164]: [1;37mutils.c[0m:[1;37m1169[0m [1;37mast_carefulwrite[0m: write() returned error: Broken pipe
    -- <SIP/alpha-centre-0000f938>AGI Script allo/get_order_state.php completed, returning 0

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:7] [1;36mNoOp[0m("[1;35mSIP/alpha-centre-0000f938[0m", "[1;35mOSTATE=10[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:8] [1;36mGotoIf[0m("[1;35mSIP/alpha-centre-0000f938[0m", "[1;35m1?To_New[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Goto (in_dial,8452900000,15)

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:15] [1;36mAnswer[0m("[1;35mSIP/alpha-centre-0000f938[0m", "[1;35m[0m") in new stack

[Ka01-centre*CLI> 
[0KAudio is at 12928
Adding codec 0x8 (alaw) to SDP

[Ka01-centre*CLI> 
[0K
<--- Reliably Transmitting (no NAT) to 192.168.2.10:5060 --->
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

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

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.2.10:5060 --->
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

<------------->
--- (10 headers 0 lines) ---

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:16] [1;36mGosub[0m("[1;35mSIP/alpha-centre-0000f938[0m", "[1;35mcall-record,,1[0m") in new stack
    -- Executing [8452900000@call-record:1] [1;36mGoto[0m("[1;35mSIP/alpha-centre-0000f938[0m", "[1;35mrecord[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Goto (call-record,8452900000,5)

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@call-record:5] [1;36mMixMonitor[0m("[1;35mSIP/alpha-centre-0000f938[0m", "[1;35m/var/spool/asterisk/monitor/record//2013/07/14/11_11_35_89616447484_8452900000_a01-centre-1373785895.63800.wav49,b[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@call-record:6] [1;36mSet[0m("[1;35mSIP/alpha-centre-0000f938[0m", "[1;35mCDR(userfield)=[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@call-record:7] [1;36mReturn[0m("[1;35mSIP/alpha-centre-0000f938[0m", "[1;35m[0m") in new stack
  == Begin MixMonitor Recording SIP/alpha-centre-0000f938

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:17] [1;36mGotoIf[0m("[1;35mSIP/alpha-centre-0000f938[0m", "[1;35m0?setprior[0m") in new stack
    -- Executing [8452900000@in_dial:18] [1;36mSet[0m("[1;35mSIP/alpha-centre-0000f938[0m", "[1;35mQUEUE_PRIO=5[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:19] [1;36mSystem[0m("[1;35mSIP/alpha-centre-0000f938[0m", "[1;35m/bin/echo "`date "+%F_%H-%M-%S"` "89616447484" <alpha-centre>" >> /tmp/calls-NEW.log[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:20] [1;36mQueue[0m("[1;35mSIP/alpha-centre-0000f938[0m", "[1;35mdispechers,tT,,,100[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Started music on hold, class 'default', on SIP/alpha-centre-0000f938

[Ka01-centre*CLI> 
[0K  == Using SIP RTP CoS mark 5

[Ka01-centre*CLI> 
[0KAudio is at 13838
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

[Ka01-centre*CLI> 
[0KReliably Transmitting (no NAT) to 192.168.0.36:5651:
INVITE sip:2038@192.168.0.36:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK127c34ae
Max-Forwards: 70
From: "89616447484" <sip:alpha-centre@192.168.2.90>;tag=as74732c7d
To: <sip:2038@192.168.0.36:5651>
Contact: <sip:alpha-centre@192.168.2.90:5060>
Call-ID: 5b3f245b5a6736613c4ba0db4b195e17@192.168.2.90:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.20.0
Date: Sun, 14 Jul 2013 07:11:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 1544252474 1544252474 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 13838 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.36:5651 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK127c34ae
From: "89616447484"<sip:alpha-centre@192.168.2.90>;tag=as74732c7d
Call-ID: 5b3f245b5a6736613c4ba0db4b195e17@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2038@192.168.0.36:5651>
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.36:5651 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK127c34ae
From: "89616447484"<sip:alpha-centre@192.168.2.90>;tag=as74732c7d
Call-ID: 5b3f245b5a6736613c4ba0db4b195e17@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2038@192.168.0.36:5651>;tag=fduvyjmj
Content-Length: 0
Warning: 399 ozsdk "Busy"

<------------->
--- (8 headers 0 lines) ---

[Ka01-centre*CLI> 
[0K    -- Got SIP response 486 "Busy Here" back from 192.168.0.36:5651

[Ka01-centre*CLI> 
[0KTransmitting (no NAT) to 192.168.0.36:5651:
ACK sip:2038@192.168.0.36:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK127c34ae
Max-Forwards: 70
From: "89616447484" <sip:alpha-centre@192.168.2.90>;tag=as74732c7d
To: <sip:2038@192.168.0.36:5651>;tag=fduvyjmj
Contact: <sip:alpha-centre@192.168.2.90:5060>
Call-ID: 5b3f245b5a6736613c4ba0db4b195e17@192.168.2.90:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.20.0
Content-Length: 0


---

[Ka01-centre*CLI> 
[0K    -- SIP/2038-0000f939 is busy

[Ka01-centre*CLI> 
[0K    -- Nobody picked up in 0 ms

[Ka01-centre*CLI> 
[0K  == Using SIP RTP CoS mark 5

[Ka01-centre*CLI> 
[0KReally destroying SIP dialog '5b3f245b5a6736613c4ba0db4b195e17@192.168.2.90:5060' Method: INVITE

[Ka01-centre*CLI> 
[0KAudio is at 19844
Adding codec 0x2 (gsm) to SDP

[Ka01-centre*CLI> 
[0KAdding non-codec 0x1 (telephone-event) to SDP

[Ka01-centre*CLI> 
[0KReliably Transmitting (no NAT) to 192.168.0.42:5651:
INVITE sip:2060@192.168.0.42:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK24692d69
Max-Forwards: 70
From: "89616447484" <sip:alpha-centre@192.168.2.90>;tag=as257c1be1
To: <sip:2060@192.168.0.42:5651>
Contact: <sip:alpha-centre@192.168.2.90:5060>
Call-ID: 5092c5ec1b06ffa3194d08cc2a9e346d@192.168.2.90:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.20.0
Date: Sun, 14 Jul 2013 07:11:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 550334780 550334780 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 19844 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.42:5651 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK24692d69
From: "89616447484"<sip:alpha-centre@192.168.2.90>;tag=as257c1be1
Call-ID: 5092c5ec1b06ffa3194d08cc2a9e346d@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2060@192.168.0.42:5651>
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.42:5651 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK24692d69
From: "89616447484"<sip:alpha-centre@192.168.2.90>;tag=as257c1be1
Call-ID: 5092c5ec1b06ffa3194d08cc2a9e346d@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2060@192.168.0.42:5651>;tag=jcnejwxu
Content-Type: application/sdp
Content-Length: 200
Contact: <sip:2060@192.168.0.42:5651>

v=0
o=- 1393899184 1393899184 IN IP4 192.168.0.42
s=Ozeki VoIP SIP SDK
c=IN IP4 192.168.0.42
t=0 0
m=audio 5650 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
--- (9 headers 9 lines) ---

[Ka01-centre*CLI> 
[0KFound RTP audio format 3
Found RTP audio format 101
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - 0x2 (gsm), peer - audio=0x2 (gsm)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x2 (gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.42:5650

[Ka01-centre*CLI> 
[0Klist_route: hop: <sip:2060@192.168.0.42:5651>

[Ka01-centre*CLI> 
[0Kset_destination: Parsing <sip:2060@192.168.0.42:5651> for address/port to send to
set_destination: set destination to 192.168.0.42:5651

[Ka01-centre*CLI> 
[0KTransmitting (no NAT) to 192.168.0.42:5651:
ACK sip:2060@192.168.0.42:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK5f10ab01
Max-Forwards: 70
From: "89616447484" <sip:alpha-centre@192.168.2.90>;tag=as257c1be1
To: <sip:2060@192.168.0.42:5651>;tag=jcnejwxu
Contact: <sip:alpha-centre@192.168.2.90:5060>
Call-ID: 5092c5ec1b06ffa3194d08cc2a9e346d@192.168.2.90:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.20.0
Content-Length: 0


---

[Ka01-centre*CLI> 
[0K    -- SIP/2060-0000f93a answered SIP/alpha-centre-0000f938

[Ka01-centre*CLI> 
[0K    -- Stopped music on hold on SIP/alpha-centre-0000f938

[Ka01-centre*CLI> 
[0KReally destroying SIP dialog '31a1fbf25b4024fd7f195aa67cfbea09@192.168.5.33:5060' Method: BYE

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:10.1.0.2:1028 --->
INVITE sip:8452440000@192.168.2.90 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.2:1028;branch=z9hG4bK1f4ec272
Max-Forwards: 70
From: "88452445099" <sip:astr-centre@10.1.0.2:1028>;tag=as6cb9abed
To: <sip:8452440000@192.168.2.90>
Contact: <sip:astr-centre@10.1.0.2:1028>
Call-ID: 75bca01a3dd655337b5e5ed716349684@192.168.5.33:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.20.0
Date: Sun, 14 Jul 2013 07:11:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 255

v=0
o=root 1687026608 1687026608 IN IP4 10.1.0.2
s=Asterisk PBX 1.8.20.0
c=IN IP4 10.1.0.2
t=0 0
m=audio 19528 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 12 lines) ---

[Ka01-centre*CLI> 
[0KSending to 10.1.0.2:1028 (NAT)
Using INVITE request as basis request - 75bca01a3dd655337b5e5ed716349684@192.168.5.33:5060
Found peer 'astr-centre' for 'astr-centre' from 10.1.0.2:1028

[Ka01-centre*CLI> 
[0K  == Using SIP RTP CoS mark 5
Found RTP audio format 3
Found RTP audio format 101
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101

[Ka01-centre*CLI> 
[0KCapabilities: us - 0x2 (gsm), peer - audio=0x2 (gsm)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x2 (gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.1.0.2:19528
Looking for 8452440000 in in_dial (domain 192.168.2.90)

[Ka01-centre*CLI> 
[0Klist_route: hop: <sip:astr-centre@10.1.0.2:1028>

<--- Transmitting (no NAT) to 10.1.0.2:1028 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.0.2:1028;branch=z9hG4bK1f4ec272;received=10.1.0.2
From: "88452445099" <sip:astr-centre@10.1.0.2:1028>;tag=as6cb9abed
To: <sip:8452440000@192.168.2.90>
Call-ID: 75bca01a3dd655337b5e5ed716349684@192.168.5.33:5060
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:8452440000@192.168.2.90:5060>
Content-Length: 0


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

[Ka01-centre*CLI> 
[0K    -- Executing [8452440000@in_dial:1] [1;36mNoOp[0m("[1;35mSIP/astr-centre-0000f93b[0m", "[1;35m<<<=== Входящий 88452445099 a01-centre-1373785898.63803 <<<===[0m") in new stack
    -- Executing [8452440000@in_dial:2] [1;36mSystem[0m("[1;35mSIP/astr-centre-0000f93b[0m", "[1;35m/bin/echo "`date "+%F_%H-%M-%S"` "88452445099" <astr-centre>" >> /tmp/calls-N.log[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452440000@in_dial:3] [1;36mGotoIf[0m("[1;35mSIP/astr-centre-0000f93b[0m", "[1;35m0?To_Old[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452440000@in_dial:4] [1;36mGotoIf[0m("[1;35mSIP/astr-centre-0000f93b[0m", "[1;35m0?blacklist[0m") in new stack
    -- Executing [8452440000@in_dial:5] [1;36mSet[0m("[1;35mSIP/astr-centre-0000f93b[0m", "[1;35mOSTATE=0[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452440000@in_dial:6] [1;36mAGI[0m("[1;35mSIP/astr-centre-0000f93b[0m", "[1;35mallo/get_order_state.php,88452445099[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Launched AGI Script /usr/share/asterisk/agi-bin/allo/get_order_state.php

[Ka01-centre*CLI> 
[0K[Jul 14 11:11:38] [0;31mERROR[0m[24189]: [1;37mutils.c[0m:[1;37m1169[0m [1;37mast_carefulwrite[0m: write() returned error: Broken pipe

[Ka01-centre*CLI> 
[0K    -- <SIP/astr-centre-0000f93b>AGI Script allo/get_order_state.php completed, returning 0

[Ka01-centre*CLI> 
[0K    -- Executing [8452440000@in_dial:7] [1;36mNoOp[0m("[1;35mSIP/astr-centre-0000f93b[0m", "[1;35mOSTATE=10[0m") in new stack
    -- Executing [8452440000@in_dial:8] [1;36mGotoIf[0m("[1;35mSIP/astr-centre-0000f93b[0m", "[1;35m1?To_New[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Goto (in_dial,8452440000,15)
    -- Executing [8452440000@in_dial:15] [1;36mAnswer[0m("[1;35mSIP/astr-centre-0000f93b[0m", "[1;35m[0m") in new stack

[Ka01-centre*CLI> 
[0KAudio is at 15822
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

[Ka01-centre*CLI> 
[0K
<--- Reliably Transmitting (no NAT) to 10.1.0.2:1028 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.0.2:1028;branch=z9hG4bK1f4ec272;received=10.1.0.2
From: "88452445099" <sip:astr-centre@10.1.0.2:1028>;tag=as6cb9abed
To: <sip:8452440000@192.168.2.90>;tag=as06923e0a
Call-ID: 75bca01a3dd655337b5e5ed716349684@192.168.5.33:5060
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:8452440000@192.168.2.90:5060>
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 781769189 781769189 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 15822 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

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

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:10.1.0.2:1028 --->
ACK sip:8452440000@192.168.2.90:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.2:1028;branch=z9hG4bK0edc2e5a
Max-Forwards: 70
From: "88452445099" <sip:astr-centre@10.1.0.2:1028>;tag=as6cb9abed
To: <sip:8452440000@192.168.2.90>;tag=as06923e0a
Contact: <sip:astr-centre@10.1.0.2:1028>
Call-ID: 75bca01a3dd655337b5e5ed716349684@192.168.5.33:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.20.0
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

[Ka01-centre*CLI> 
[0K    -- Executing [8452440000@in_dial:16] [1;36mGosub[0m("[1;35mSIP/astr-centre-0000f93b[0m", "[1;35mcall-record,,1[0m") in new stack
    -- Executing [8452440000@call-record:1] [1;36mGoto[0m("[1;35mSIP/astr-centre-0000f93b[0m", "[1;35mrecord[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Goto (call-record,8452440000,5)

[Ka01-centre*CLI> 
[0K    -- Executing [8452440000@call-record:5] [1;36mMixMonitor[0m("[1;35mSIP/astr-centre-0000f93b[0m", "[1;35m/var/spool/asterisk/monitor/record//2013/07/14/11_11_38_88452445099_8452440000_a01-centre-1373785898.63803.wav49,b[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452440000@call-record:6] [1;36mSet[0m("[1;35mSIP/astr-centre-0000f93b[0m", "[1;35mCDR(userfield)=[0m") in new stack

[Ka01-centre*CLI> 
[0K  == Begin MixMonitor Recording SIP/astr-centre-0000f93b

[Ka01-centre*CLI> 
[0K    -- Executing [8452440000@call-record:7] [1;36mReturn[0m("[1;35mSIP/astr-centre-0000f93b[0m", "[1;35m[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452440000@in_dial:17] [1;36mGotoIf[0m("[1;35mSIP/astr-centre-0000f93b[0m", "[1;35m0?setprior[0m") in new stack
    -- Executing [8452440000@in_dial:18] [1;36mSet[0m("[1;35mSIP/astr-centre-0000f93b[0m", "[1;35mQUEUE_PRIO=5[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452440000@in_dial:19] [1;36mSystem[0m("[1;35mSIP/astr-centre-0000f93b[0m", "[1;35m/bin/echo "`date "+%F_%H-%M-%S"` "88452445099" <astr-centre>" >> /tmp/calls-NEW.log[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452440000@in_dial:20] [1;36mQueue[0m("[1;35mSIP/astr-centre-0000f93b[0m", "[1;35mdispechers,tT,,,100[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Started music on hold, class 'default', on SIP/astr-centre-0000f93b

[Ka01-centre*CLI> 
[0K  == Using SIP RTP CoS mark 5

[Ka01-centre*CLI> 
[0KAudio is at 18724
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

[Ka01-centre*CLI> 
[0KReliably Transmitting (no NAT) to 192.168.0.39:5651:
INVITE sip:2068@192.168.0.39:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK04bf1a8d
Max-Forwards: 70
From: "88452445099" <sip:astr-centre@192.168.2.90>;tag=as6ae64b9b
To: <sip:2068@192.168.0.39:5651>
Contact: <sip:astr-centre@192.168.2.90:5060>
Call-ID: 570229457a40788c7384dec023634756@192.168.2.90:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.20.0
Date: Sun, 14 Jul 2013 07:11:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 1527720500 1527720500 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 18724 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.39:5651 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK04bf1a8d
From: "88452445099"<sip:astr-centre@192.168.2.90>;tag=as6ae64b9b
Call-ID: 570229457a40788c7384dec023634756@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2068@192.168.0.39:5651>
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.39:5651 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK04bf1a8d
From: "88452445099"<sip:astr-centre@192.168.2.90>;tag=as6ae64b9b
Call-ID: 570229457a40788c7384dec023634756@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2068@192.168.0.39:5651>;tag=kxtuacnx
Content-Length: 0
Warning: 399 ozsdk "Busy"

<------------->
--- (8 headers 0 lines) ---

[Ka01-centre*CLI> 
[0K    -- Got SIP response 486 "Busy Here" back from 192.168.0.39:5651

[Ka01-centre*CLI> 
[0KTransmitting (no NAT) to 192.168.0.39:5651:
ACK sip:2068@192.168.0.39:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK04bf1a8d
Max-Forwards: 70
From: "88452445099" <sip:astr-centre@192.168.2.90>;tag=as6ae64b9b
To: <sip:2068@192.168.0.39:5651>;tag=kxtuacnx
Contact: <sip:astr-centre@192.168.2.90:5060>
Call-ID: 570229457a40788c7384dec023634756@192.168.2.90:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.20.0
Content-Length: 0


---

[Ka01-centre*CLI> 
[0K    -- SIP/2068-0000f93c is busy

[Ka01-centre*CLI> 
[0K    -- Nobody picked up in 0 ms

[Ka01-centre*CLI> 
[0K  == Using SIP RTP CoS mark 5

[Ka01-centre*CLI> 
[0KReally destroying SIP dialog '570229457a40788c7384dec023634756@192.168.2.90:5060' Method: INVITE

[Ka01-centre*CLI> 
[0KAudio is at 16688

[Ka01-centre*CLI> 
[0KAdding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

[Ka01-centre*CLI> 
[0KReliably Transmitting (no NAT) to 192.168.0.38:5651:
INVITE sip:2085@192.168.0.38:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK370626c3
Max-Forwards: 70
From: "88452445099" <sip:astr-centre@192.168.2.90>;tag=as070a7c18
To: <sip:2085@192.168.0.38:5651>
Contact: <sip:astr-centre@192.168.2.90:5060>
Call-ID: 141717f518053a6d5c3ccfef4c4ec469@192.168.2.90:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.20.0
Date: Sun, 14 Jul 2013 07:11:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 1455895830 1455895830 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 16688 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.38:5651 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK370626c3
From: "88452445099"<sip:astr-centre@192.168.2.90>;tag=as070a7c18
Call-ID: 141717f518053a6d5c3ccfef4c4ec469@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2085@192.168.0.38:5651>
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

[Ka01-centre*CLI> 
[0KReally destroying SIP dialog 'exoonxncmpvsjlaaupnbermcbwlsudtrlcprlakuwnbmirfacx' Method: REGISTER

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.38:5651 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK370626c3
From: "88452445099"<sip:astr-centre@192.168.2.90>;tag=as070a7c18
Call-ID: 141717f518053a6d5c3ccfef4c4ec469@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2085@192.168.0.38:5651>;tag=gulqivbw
Content-Length: 0
Warning: 399 ozsdk "User reject"

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

[Ka01-centre*CLI> 
[0K--- (8 headers 0 lines) ---
    -- Got SIP response 486 "Busy Here" back from 192.168.0.38:5651
Transmitting (no NAT) to 192.168.0.38:5651:
ACK sip:2085@192.168.0.38:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK370626c3
Max-Forwards: 70
From: "88452445099" <sip:astr-centre@192.168.2.90>;tag=as070a7c18
To: <sip:2085@192.168.0.38:5651>;tag=gulqivbw
Contact: <sip:astr-centre@192.168.2.90:5060>
Call-ID: 141717f518053a6d5c3ccfef4c4ec469@192.168.2.90:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.20.0
Content-Length: 0


---

[Ka01-centre*CLI> 
[0K    -- SIP/2085-0000f93d is busy
    -- Nobody picked up in 1000 ms

[Ka01-centre*CLI> 
[0K  == Using SIP RTP CoS mark 5

[Ka01-centre*CLI> 
[0KReally destroying SIP dialog '141717f518053a6d5c3ccfef4c4ec469@192.168.2.90:5060' Method: INVITE

[Ka01-centre*CLI> 
[0KAudio is at 19438
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.0.36:5651:
INVITE sip:2038@192.168.0.36:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK13b61605
Max-Forwards: 70
From: "88452445099" <sip:astr-centre@192.168.2.90>;tag=as3a3d0065
To: <sip:2038@192.168.0.36:5651>
Contact: <sip:astr-centre@192.168.2.90:5060>
Call-ID: 6e6410d21d57204e442539754fb5941f@192.168.2.90:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.20.0
Date: Sun, 14 Jul 2013 07:11:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 2074640469 2074640469 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 19438 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.36:5651 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK13b61605
From: "88452445099"<sip:astr-centre@192.168.2.90>;tag=as3a3d0065
Call-ID: 6e6410d21d57204e442539754fb5941f@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2038@192.168.0.36:5651>
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.36:5651 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK13b61605
From: "88452445099"<sip:astr-centre@192.168.2.90>;tag=as3a3d0065
Call-ID: 6e6410d21d57204e442539754fb5941f@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2038@192.168.0.36:5651>;tag=oftqmfvn
Content-Length: 0
Warning: 399 ozsdk "Busy"

<------------->
--- (8 headers 0 lines) ---
    -- Got SIP response 486 "Busy Here" back from 192.168.0.36:5651
Transmitting (no NAT) to 192.168.0.36:5651:
ACK sip:2038@192.168.0.36:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK13b61605
Max-Forwards: 70
From: "88452445099" <sip:astr-centre@192.168.2.90>;tag=as3a3d0065
To: <sip:2038@192.168.0.36:5651>;tag=oftqmfvn
Contact: <sip:astr-centre@192.168.2.90:5060>
Call-ID: 6e6410d21d57204e442539754fb5941f@192.168.2.90:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.20.0
Content-Length: 0


---

[Ka01-centre*CLI> 
[0K    -- SIP/2038-0000f93e is busy

[Ka01-centre*CLI> 
[0K    -- Nobody picked up in 0 ms

[Ka01-centre*CLI> 
[0K  == Using SIP RTP CoS mark 5

[Ka01-centre*CLI> 
[0KReally destroying SIP dialog '6e6410d21d57204e442539754fb5941f@192.168.2.90:5060' Method: INVITE
Audio is at 18400

[Ka01-centre*CLI> 
[0KAdding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

[Ka01-centre*CLI> 
[0KReliably Transmitting (no NAT) to 192.168.0.42:5651:
INVITE sip:2060@192.168.0.42:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK5b7e0bf5
Max-Forwards: 70
From: "88452445099" <sip:astr-centre@192.168.2.90>;tag=as4e2d59c4
To: <sip:2060@192.168.0.42:5651>
Contact: <sip:astr-centre@192.168.2.90:5060>
Call-ID: 4e0f992b09a0eed0411901857e9d5215@192.168.2.90:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.20.0
Date: Sun, 14 Jul 2013 07:11:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 474411179 474411179 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 18400 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.42:5651 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK5b7e0bf5
From: "88452445099"<sip:astr-centre@192.168.2.90>;tag=as4e2d59c4
Call-ID: 4e0f992b09a0eed0411901857e9d5215@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2060@192.168.0.42:5651>
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.42:5651 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK5b7e0bf5
From: "88452445099"<sip:astr-centre@192.168.2.90>;tag=as4e2d59c4
Call-ID: 4e0f992b09a0eed0411901857e9d5215@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2060@192.168.0.42:5651>;tag=ynerfdgf
Content-Length: 0
Warning: 399 ozsdk "User reject"

<------------->
--- (8 headers 0 lines) ---
    -- Got SIP response 486 "Busy Here" back from 192.168.0.42:5651
Transmitting (no NAT) to 192.168.0.42:5651:
ACK sip:2060@192.168.0.42:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK5b7e0bf5
Max-Forwards: 70
From: "88452445099" <sip:astr-centre@192.168.2.90>;tag=as4e2d59c4
To: <sip:2060@192.168.0.42:5651>;tag=ynerfdgf
Contact: <sip:astr-centre@192.168.2.90:5060>
Call-ID: 4e0f992b09a0eed0411901857e9d5215@192.168.2.90:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.20.0
Content-Length: 0


---

[Ka01-centre*CLI> 
[0K    -- SIP/2060-0000f93f is busy

[Ka01-centre*CLI> 
[0K    -- Nobody picked up in 2000 ms

[Ka01-centre*CLI> 
[0KReally destroying SIP dialog '4e0f992b09a0eed0411901857e9d5215@192.168.2.90:5060' Method: INVITE

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.2.10:5060 --->
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
<------------->
--- (14 headers 9 lines) ---

[Ka01-centre*CLI> 
[0KSending to 192.168.2.10:5060 (NAT)
Using INVITE request as basis request - 5a082ca80d6daaa3128cc4dd5e8af1af@192.168.2.10

[Ka01-centre*CLI> 
[0KFound peer 'alpha-centre' for 'alpha-centre' from 192.168.2.10:5060

[Ka01-centre*CLI> 
[0K  == Using SIP RTP CoS mark 5

[Ka01-centre*CLI> 
[0KFound RTP audio format 8

[Ka01-centre*CLI> 
[0KFound audio description format PCMA for ID 8
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.2.10:18562

[Ka01-centre*CLI> 
[0KLooking for 8452900000 in in_dial (domain 192.168.2.90)

[Ka01-centre*CLI> 
[0Klist_route: hop: <sip:alpha-centre@192.168.2.10>

[Ka01-centre*CLI> 
[0K
<--- Transmitting (no NAT) to 192.168.2.10:5060 --->
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


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

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:1] [1;36mNoOp[0m("[1;35mSIP/alpha-centre-0000f940[0m", "[1;35m<<<=== Входящий 89372542388 a01-centre-1373785904.63808 <<<===[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:2] [1;36mSystem[0m("[1;35mSIP/alpha-centre-0000f940[0m", "[1;35m/bin/echo "`date "+%F_%H-%M-%S"` "89372542388" <alpha-centre>" >> /tmp/calls-N.log[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:3] [1;36mGotoIf[0m("[1;35mSIP/alpha-centre-0000f940[0m", "[1;35m0?To_Old[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:4] [1;36mGotoIf[0m("[1;35mSIP/alpha-centre-0000f940[0m", "[1;35m0?blacklist[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:5] [1;36mSet[0m("[1;35mSIP/alpha-centre-0000f940[0m", "[1;35mOSTATE=0[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:6] [1;36mAGI[0m("[1;35mSIP/alpha-centre-0000f940[0m", "[1;35mallo/get_order_state.php,89372542388[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Launched AGI Script /usr/share/asterisk/agi-bin/allo/get_order_state.php

[Ka01-centre*CLI> 
[0K[Jul 14 11:11:44] [0;31mERROR[0m[24215]: [1;37mutils.c[0m:[1;37m1169[0m [1;37mast_carefulwrite[0m: write() returned error: Broken pipe
    -- <SIP/alpha-centre-0000f940>AGI Script allo/get_order_state.php completed, returning 0

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:7] [1;36mNoOp[0m("[1;35mSIP/alpha-centre-0000f940[0m", "[1;35mOSTATE=10[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:8] [1;36mGotoIf[0m("[1;35mSIP/alpha-centre-0000f940[0m", "[1;35m1?To_New[0m") in new stack
    -- Goto (in_dial,8452900000,15)
    -- Executing [8452900000@in_dial:15] [1;36mAnswer[0m("[1;35mSIP/alpha-centre-0000f940[0m", "[1;35m[0m") in new stack

[Ka01-centre*CLI> 
[0KAudio is at 19836
Adding codec 0x8 (alaw) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.2.10:5060 --->
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

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

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.2.10:5060 --->
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

<------------->
--- (10 headers 0 lines) ---

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:16] [1;36mGosub[0m("[1;35mSIP/alpha-centre-0000f940[0m", "[1;35mcall-record,,1[0m") in new stack
    -- Executing [8452900000@call-record:1] [1;36mGoto[0m("[1;35mSIP/alpha-centre-0000f940[0m", "[1;35mrecord[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Goto (call-record,8452900000,5)

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@call-record:5] [1;36mMixMonitor[0m("[1;35mSIP/alpha-centre-0000f940[0m", "[1;35m/var/spool/asterisk/monitor/record//2013/07/14/11_11_44_89372542388_8452900000_a01-centre-1373785904.63808.wav49,b[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@call-record:6] [1;36mSet[0m("[1;35mSIP/alpha-centre-0000f940[0m", "[1;35mCDR(userfield)=[0m") in new stack
  == Begin MixMonitor Recording SIP/alpha-centre-0000f940
    -- Executing [8452900000@call-record:7] [1;36mReturn[0m("[1;35mSIP/alpha-centre-0000f940[0m", "[1;35m[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:17] [1;36mGotoIf[0m("[1;35mSIP/alpha-centre-0000f940[0m", "[1;35m0?setprior[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:18] [1;36mSet[0m("[1;35mSIP/alpha-centre-0000f940[0m", "[1;35mQUEUE_PRIO=5[0m") in new stack
    -- Executing [8452900000@in_dial:19] [1;36mSystem[0m("[1;35mSIP/alpha-centre-0000f940[0m", "[1;35m/bin/echo "`date "+%F_%H-%M-%S"` "89372542388" <alpha-centre>" >> /tmp/calls-NEW.log[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:20] [1;36mQueue[0m("[1;35mSIP/alpha-centre-0000f940[0m", "[1;35mdispechers,tT,,,100[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Started music on hold, class 'default', on SIP/alpha-centre-0000f940

[Ka01-centre*CLI> 
[0K  == Using SIP RTP CoS mark 5

[Ka01-centre*CLI> 
[0KAudio is at 10616
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

[Ka01-centre*CLI> 
[0KReliably Transmitting (no NAT) to 192.168.0.36:5651:
INVITE sip:2038@192.168.0.36:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK26360260
Max-Forwards: 70
From: "89372542388" <sip:alpha-centre@192.168.2.90>;tag=as355f8722
To: <sip:2038@192.168.0.36:5651>
Contact: <sip:alpha-centre@192.168.2.90:5060>
Call-ID: 5d9b28a74921aae94b1a16f40448f825@192.168.2.90:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.20.0
Date: Sun, 14 Jul 2013 07:11:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 1464756186 1464756186 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 10616 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.36:5651 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK26360260
From: "89372542388"<sip:alpha-centre@192.168.2.90>;tag=as355f8722
Call-ID: 5d9b28a74921aae94b1a16f40448f825@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2038@192.168.0.36:5651>
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.36:5651 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK26360260
From: "89372542388"<sip:alpha-centre@192.168.2.90>;tag=as355f8722
Call-ID: 5d9b28a74921aae94b1a16f40448f825@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2038@192.168.0.36:5651>;tag=lapdbwhc
Content-Length: 0
Warning: 399 ozsdk "Busy"

<------------->
--- (8 headers 0 lines) ---

[Ka01-centre*CLI> 
[0K    -- Got SIP response 486 "Busy Here" back from 192.168.0.36:5651

[Ka01-centre*CLI> 
[0KTransmitting (no NAT) to 192.168.0.36:5651:
ACK sip:2038@192.168.0.36:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK26360260
Max-Forwards: 70
From: "89372542388" <sip:alpha-centre@192.168.2.90>;tag=as355f8722
To: <sip:2038@192.168.0.36:5651>;tag=lapdbwhc
Contact: <sip:alpha-centre@192.168.2.90:5060>
Call-ID: 5d9b28a74921aae94b1a16f40448f825@192.168.2.90:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.20.0
Content-Length: 0


---

[Ka01-centre*CLI> 
[0K    -- SIP/2038-0000f941 is busy

[Ka01-centre*CLI> 
[0K    -- Nobody picked up in 0 ms

[Ka01-centre*CLI> 
[0K  == Using SIP RTP CoS mark 5

[Ka01-centre*CLI> 
[0KReally destroying SIP dialog '5d9b28a74921aae94b1a16f40448f825@192.168.2.90:5060' Method: INVITE

[Ka01-centre*CLI> 
[0KAudio is at 11498

[Ka01-centre*CLI> 
[0KAdding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

[Ka01-centre*CLI> 
[0KReliably Transmitting (no NAT) to 192.168.0.42:5651:
INVITE sip:2060@192.168.0.42:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK6a5ac93e
Max-Forwards: 70
From: "89372542388" <sip:alpha-centre@192.168.2.90>;tag=as0eaf07fd
To: <sip:2060@192.168.0.42:5651>
Contact: <sip:alpha-centre@192.168.2.90:5060>
Call-ID: 63e2ce6054b2e2477f401c387bbe43bd@192.168.2.90:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.20.0
Date: Sun, 14 Jul 2013 07:11:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 1786419703 1786419703 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 11498 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.42:5651 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK6a5ac93e
From: "89372542388"<sip:alpha-centre@192.168.2.90>;tag=as0eaf07fd
Call-ID: 63e2ce6054b2e2477f401c387bbe43bd@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2060@192.168.0.42:5651>
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.2.10:5060 --->
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
<------------->
--- (14 headers 9 lines) ---

[Ka01-centre*CLI> 
[0KSending to 192.168.2.10:5060 (NAT)
Using INVITE request as basis request - 0e7f74aa1f6d814863ccc98c132f17ed@192.168.2.10
Found peer 'alpha-centre' for 'alpha-centre' from 192.168.2.10:5060

[Ka01-centre*CLI> 
[0K  == Using SIP RTP CoS mark 5

[Ka01-centre*CLI> 
[0KFound RTP audio format 8
Found audio description format PCMA for ID 8
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.2.10:12782
Looking for 8452900000 in in_dial (domain 192.168.2.90)

[Ka01-centre*CLI> 
[0Klist_route: hop: <sip:alpha-centre@192.168.2.10>

<--- Transmitting (no NAT) to 192.168.2.10:5060 --->
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


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

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:1] [1;36mNoOp[0m("[1;35mSIP/alpha-centre-0000f943[0m", "[1;35m<<<=== Входящий 89172011440 a01-centre-1373785905.63811 <<<===[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:2] [1;36mSystem[0m("[1;35mSIP/alpha-centre-0000f943[0m", "[1;35m/bin/echo "`date "+%F_%H-%M-%S"` "89172011440" <alpha-centre>" >> /tmp/calls-N.log[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:3] [1;36mGotoIf[0m("[1;35mSIP/alpha-centre-0000f943[0m", "[1;35m0?To_Old[0m") in new stack
    -- Executing [8452900000@in_dial:4] [1;36mGotoIf[0m("[1;35mSIP/alpha-centre-0000f943[0m", "[1;35m0?blacklist[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:5] [1;36mSet[0m("[1;35mSIP/alpha-centre-0000f943[0m", "[1;35mOSTATE=0[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:6] [1;36mAGI[0m("[1;35mSIP/alpha-centre-0000f943[0m", "[1;35mallo/get_order_state.php,89172011440[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Launched AGI Script /usr/share/asterisk/agi-bin/allo/get_order_state.php

[Ka01-centre*CLI> 
[0K[Jul 14 11:11:45] [0;31mERROR[0m[24226]: [1;37mutils.c[0m:[1;37m1169[0m [1;37mast_carefulwrite[0m: write() returned error: Broken pipe
    -- <SIP/alpha-centre-0000f943>AGI Script allo/get_order_state.php completed, returning 0

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:7] [1;36mNoOp[0m("[1;35mSIP/alpha-centre-0000f943[0m", "[1;35mOSTATE=10[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:8] [1;36mGotoIf[0m("[1;35mSIP/alpha-centre-0000f943[0m", "[1;35m1?To_New[0m") in new stack
    -- Goto (in_dial,8452900000,15)
    -- Executing [8452900000@in_dial:15] [1;36mAnswer[0m("[1;35mSIP/alpha-centre-0000f943[0m", "[1;35m[0m") in new stack

[Ka01-centre*CLI> 
[0KAudio is at 10086
Adding codec 0x8 (alaw) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.2.10:5060 --->
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

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

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.2.10:5060 --->
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

<------------->
--- (10 headers 0 lines) ---

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:16] [1;36mGosub[0m("[1;35mSIP/alpha-centre-0000f943[0m", "[1;35mcall-record,,1[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@call-record:1] [1;36mGoto[0m("[1;35mSIP/alpha-centre-0000f943[0m", "[1;35mrecord[0m") in new stack
    -- Goto (call-record,8452900000,5)

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@call-record:5] [1;36mMixMonitor[0m("[1;35mSIP/alpha-centre-0000f943[0m", "[1;35m/var/spool/asterisk/monitor/record//2013/07/14/11_11_45_89172011440_8452900000_a01-centre-1373785905.63811.wav49,b[0m") in new stack

[Ka01-centre*CLI> 
[0K  == Begin MixMonitor Recording SIP/alpha-centre-0000f943
    -- Executing [8452900000@call-record:6] [1;36mSet[0m("[1;35mSIP/alpha-centre-0000f943[0m", "[1;35mCDR(userfield)=[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@call-record:7] [1;36mReturn[0m("[1;35mSIP/alpha-centre-0000f943[0m", "[1;35m[0m") in new stack
    -- Executing [8452900000@in_dial:17] [1;36mGotoIf[0m("[1;35mSIP/alpha-centre-0000f943[0m", "[1;35m0?setprior[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:18] [1;36mSet[0m("[1;35mSIP/alpha-centre-0000f943[0m", "[1;35mQUEUE_PRIO=5[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:19] [1;36mSystem[0m("[1;35mSIP/alpha-centre-0000f943[0m", "[1;35m/bin/echo "`date "+%F_%H-%M-%S"` "89172011440" <alpha-centre>" >> /tmp/calls-NEW.log[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:20] [1;36mQueue[0m("[1;35mSIP/alpha-centre-0000f943[0m", "[1;35mdispechers,tT,,,100[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Started music on hold, class 'default', on SIP/alpha-centre-0000f943

[Ka01-centre*CLI> 
[0K  == Using SIP RTP CoS mark 5

[Ka01-centre*CLI> 
[0KAudio is at 16230

[Ka01-centre*CLI> 
[0KAdding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

[Ka01-centre*CLI> 
[0KReliably Transmitting (no NAT) to 192.168.0.36:5651:
INVITE sip:2038@192.168.0.36:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK748af9ae
Max-Forwards: 70
From: "89172011440" <sip:alpha-centre@192.168.2.90>;tag=as19b0d5eb
To: <sip:2038@192.168.0.36:5651>
Contact: <sip:alpha-centre@192.168.2.90:5060>
Call-ID: 3e02a34c24b3b2e32bc55ee85c50c841@192.168.2.90:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.20.0
Date: Sun, 14 Jul 2013 07:11:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 909633176 909633176 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 16230 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.36:5651 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK748af9ae
From: "89172011440"<sip:alpha-centre@192.168.2.90>;tag=as19b0d5eb
Call-ID: 3e02a34c24b3b2e32bc55ee85c50c841@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2038@192.168.0.36:5651>
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.36:5651 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK748af9ae
From: "89172011440"<sip:alpha-centre@192.168.2.90>;tag=as19b0d5eb
Call-ID: 3e02a34c24b3b2e32bc55ee85c50c841@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2038@192.168.0.36:5651>;tag=sbrlqctp
Content-Length: 0
Warning: 399 ozsdk "Busy"

<------------->
--- (8 headers 0 lines) ---

[Ka01-centre*CLI> 
[0K    -- Got SIP response 486 "Busy Here" back from 192.168.0.36:5651

[Ka01-centre*CLI> 
[0KTransmitting (no NAT) to 192.168.0.36:5651:
ACK sip:2038@192.168.0.36:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK748af9ae
Max-Forwards: 70
From: "89172011440" <sip:alpha-centre@192.168.2.90>;tag=as19b0d5eb
To: <sip:2038@192.168.0.36:5651>;tag=sbrlqctp
Contact: <sip:alpha-centre@192.168.2.90:5060>
Call-ID: 3e02a34c24b3b2e32bc55ee85c50c841@192.168.2.90:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.20.0
Content-Length: 0


---

[Ka01-centre*CLI> 
[0K    -- SIP/2038-0000f944 is busy

[Ka01-centre*CLI> 
[0K    -- Nobody picked up in 0 ms

[Ka01-centre*CLI> 
[0K  == Using SIP RTP CoS mark 5

[Ka01-centre*CLI> 
[0KReally destroying SIP dialog '3e02a34c24b3b2e32bc55ee85c50c841@192.168.2.90:5060' Method: INVITE

[Ka01-centre*CLI> 
[0KAudio is at 10528
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

[Ka01-centre*CLI> 
[0KReliably Transmitting (no NAT) to 192.168.0.42:5651:
INVITE sip:2060@192.168.0.42:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK60468c7e
Max-Forwards: 70
From: "89172011440" <sip:alpha-centre@192.168.2.90>;tag=as1cafa869
To: <sip:2060@192.168.0.42:5651>
Contact: <sip:alpha-centre@192.168.2.90:5060>
Call-ID: 3a1ce03b22d49a4d622c5675314dd2a4@192.168.2.90:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.20.0
Date: Sun, 14 Jul 2013 07:11:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 1785388200 1785388200 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 10528 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

[Ka01-centre*CLI> 
[0KRetransmitting #1 (no NAT) to 192.168.0.42:5651:
INVITE sip:2060@192.168.0.42:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK60468c7e
Max-Forwards: 70
From: "89172011440" <sip:alpha-centre@192.168.2.90>;tag=as1cafa869
To: <sip:2060@192.168.0.42:5651>
Contact: <sip:alpha-centre@192.168.2.90:5060>
Call-ID: 3a1ce03b22d49a4d622c5675314dd2a4@192.168.2.90:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.20.0
Date: Sun, 14 Jul 2013 07:11:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 1785388200 1785388200 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 10528 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.42:5651 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK6a5ac93e
From: "89372542388"<sip:alpha-centre@192.168.2.90>;tag=as0eaf07fd
Call-ID: 63e2ce6054b2e2477f401c387bbe43bd@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2060@192.168.0.42:5651>;tag=lcqpxkpc
Content-Length: 0
Warning: 399 ozsdk "User reject"

<------------->
--- (8 headers 0 lines) ---

[Ka01-centre*CLI> 
[0K    -- Got SIP response 486 "Busy Here" back from 192.168.0.42:5651

[Ka01-centre*CLI> 
[0KTransmitting (no NAT) to 192.168.0.42:5651:
ACK sip:2060@192.168.0.42:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK6a5ac93e
Max-Forwards: 70
From: "89372542388" <sip:alpha-centre@192.168.2.90>;tag=as0eaf07fd
To: <sip:2060@192.168.0.42:5651>;tag=lcqpxkpc
Contact: <sip:alpha-centre@192.168.2.90:5060>
Call-ID: 63e2ce6054b2e2477f401c387bbe43bd@192.168.2.90:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.20.0
Content-Length: 0


---

[Ka01-centre*CLI> 
[0K    -- SIP/2060-0000f942 is busy

[Ka01-centre*CLI> 
[0K    -- Nobody picked up in 1000 ms

[Ka01-centre*CLI> 
[0K  == Using SIP RTP CoS mark 5

[Ka01-centre*CLI> 
[0KReally destroying SIP dialog '63e2ce6054b2e2477f401c387bbe43bd@192.168.2.90:5060' Method: INVITE

[Ka01-centre*CLI> 
[0KAudio is at 19964
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

[Ka01-centre*CLI> 
[0KReliably Transmitting (no NAT) to 192.168.0.39:5651:
INVITE sip:2068@192.168.0.39:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK43679d0d
Max-Forwards: 70
From: "89372542388" <sip:alpha-centre@192.168.2.90>;tag=as3b385438
To: <sip:2068@192.168.0.39:5651>
Contact: <sip:alpha-centre@192.168.2.90:5060>
Call-ID: 7faa264d3bc143392b159bdb4c8e2d18@192.168.2.90:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.20.0
Date: Sun, 14 Jul 2013 07:11:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 1059019166 1059019166 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 19964 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.42:5651 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK60468c7e
From: "89172011440"<sip:alpha-centre@192.168.2.90>;tag=as1cafa869
Call-ID: 3a1ce03b22d49a4d622c5675314dd2a4@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2060@192.168.0.42:5651>
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.39:5651 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK43679d0d
From: "89372542388"<sip:alpha-centre@192.168.2.90>;tag=as3b385438
Call-ID: 7faa264d3bc143392b159bdb4c8e2d18@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2068@192.168.0.39:5651>
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.39:5651 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK43679d0d
From: "89372542388"<sip:alpha-centre@192.168.2.90>;tag=as3b385438
Call-ID: 7faa264d3bc143392b159bdb4c8e2d18@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2068@192.168.0.39:5651>;tag=fxlvbkkk
Content-Length: 0
Warning: 399 ozsdk "Busy"

<------------->
--- (8 headers 0 lines) ---
    -- Got SIP response 486 "Busy Here" back from 192.168.0.39:5651

[Ka01-centre*CLI> 
[0KTransmitting (no NAT) to 192.168.0.39:5651:
ACK sip:2068@192.168.0.39:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK43679d0d
Max-Forwards: 70
From: "89372542388" <sip:alpha-centre@192.168.2.90>;tag=as3b385438
To: <sip:2068@192.168.0.39:5651>;tag=fxlvbkkk
Contact: <sip:alpha-centre@192.168.2.90:5060>
Call-ID: 7faa264d3bc143392b159bdb4c8e2d18@192.168.2.90:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.20.0
Content-Length: 0


---

[Ka01-centre*CLI> 
[0K    -- SIP/2068-0000f946 is busy
    -- Nobody picked up in 0 ms

[Ka01-centre*CLI> 
[0K  == Using SIP RTP CoS mark 5

[Ka01-centre*CLI> 
[0KReally destroying SIP dialog '7faa264d3bc143392b159bdb4c8e2d18@192.168.2.90:5060' Method: INVITE

[Ka01-centre*CLI> 
[0KAudio is at 17748
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

[Ka01-centre*CLI> 
[0KReliably Transmitting (no NAT) to 192.168.0.38:5651:
INVITE sip:2085@192.168.0.38:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK66103e56
Max-Forwards: 70
From: "89372542388" <sip:alpha-centre@192.168.2.90>;tag=as48533cf8
To: <sip:2085@192.168.0.38:5651>
Contact: <sip:alpha-centre@192.168.2.90:5060>
Call-ID: 1ab1c531084702b23ae602c83767f03e@192.168.2.90:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.20.0
Date: Sun, 14 Jul 2013 07:11:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 1737421027 1737421027 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 17748 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.38:5651 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK66103e56
From: "89372542388"<sip:alpha-centre@192.168.2.90>;tag=as48533cf8
Call-ID: 1ab1c531084702b23ae602c83767f03e@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2085@192.168.0.38:5651>
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.42:5651 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK6a5ac93e
From: "89372542388"<sip:alpha-centre@192.168.2.90>;tag=as0eaf07fd
Call-ID: 63e2ce6054b2e2477f401c387bbe43bd@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2060@192.168.0.42:5651>;tag=lcqpxkpc
Content-Length: 0
Warning: 399 ozsdk "User reject"

<------------->
--- (8 headers 0 lines) ---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.42:5651 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK60468c7e
From: "89172011440"<sip:alpha-centre@192.168.2.90>;tag=as1cafa869
Call-ID: 3a1ce03b22d49a4d622c5675314dd2a4@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2060@192.168.0.42:5651>;tag=ngqeyyfo
Content-Length: 0
Warning: 399 ozsdk "User reject"

<------------->
--- (8 headers 0 lines) ---
    -- Got SIP response 486 "Busy Here" back from 192.168.0.42:5651

[Ka01-centre*CLI> 
[0KTransmitting (no NAT) to 192.168.0.42:5651:
ACK sip:2060@192.168.0.42:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK60468c7e
Max-Forwards: 70
From: "89172011440" <sip:alpha-centre@192.168.2.90>;tag=as1cafa869
To: <sip:2060@192.168.0.42:5651>;tag=ngqeyyfo
Contact: <sip:alpha-centre@192.168.2.90:5060>
Call-ID: 3a1ce03b22d49a4d622c5675314dd2a4@192.168.2.90:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.20.0
Content-Length: 0


---

[Ka01-centre*CLI> 
[0K    -- SIP/2060-0000f945 is busy

[Ka01-centre*CLI> 
[0K    -- Nobody picked up in 2000 ms

[Ka01-centre*CLI> 
[0K  == Using SIP RTP CoS mark 5

[Ka01-centre*CLI> 
[0KReally destroying SIP dialog '3a1ce03b22d49a4d622c5675314dd2a4@192.168.2.90:5060' Method: INVITE

[Ka01-centre*CLI> 
[0KAudio is at 18044
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

[Ka01-centre*CLI> 
[0KReliably Transmitting (no NAT) to 192.168.0.39:5651:
INVITE sip:2068@192.168.0.39:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK380c1513
Max-Forwards: 70
From: "89172011440" <sip:alpha-centre@192.168.2.90>;tag=as488d1998
To: <sip:2068@192.168.0.39:5651>
Contact: <sip:alpha-centre@192.168.2.90:5060>
Call-ID: 529bc2d62ea38519113519fa0ec97761@192.168.2.90:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.20.0
Date: Sun, 14 Jul 2013 07:11:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 1004798375 1004798375 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 18044 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.39:5651 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK380c1513
From: "89172011440"<sip:alpha-centre@192.168.2.90>;tag=as488d1998
Call-ID: 529bc2d62ea38519113519fa0ec97761@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2068@192.168.0.39:5651>
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.39:5651 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK380c1513
From: "89172011440"<sip:alpha-centre@192.168.2.90>;tag=as488d1998
Call-ID: 529bc2d62ea38519113519fa0ec97761@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2068@192.168.0.39:5651>;tag=kjmkyxnj
Content-Length: 0
Warning: 399 ozsdk "Busy"

<------------->
--- (8 headers 0 lines) ---

[Ka01-centre*CLI> 
[0K    -- Got SIP response 486 "Busy Here" back from 192.168.0.39:5651

[Ka01-centre*CLI> 
[0KTransmitting (no NAT) to 192.168.0.39:5651:
ACK sip:2068@192.168.0.39:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK380c1513
Max-Forwards: 70
From: "89172011440" <sip:alpha-centre@192.168.2.90>;tag=as488d1998
To: <sip:2068@192.168.0.39:5651>;tag=kjmkyxnj
Contact: <sip:alpha-centre@192.168.2.90:5060>
Call-ID: 529bc2d62ea38519113519fa0ec97761@192.168.2.90:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.20.0
Content-Length: 0


---

[Ka01-centre*CLI> 
[0K    -- SIP/2068-0000f948 is busy

[Ka01-centre*CLI> 
[0K    -- Nobody picked up in 0 ms

[Ka01-centre*CLI> 
[0K  == Using SIP RTP CoS mark 5

[Ka01-centre*CLI> 
[0KReally destroying SIP dialog '529bc2d62ea38519113519fa0ec97761@192.168.2.90:5060' Method: INVITE

[Ka01-centre*CLI> 
[0KAudio is at 19640
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

[Ka01-centre*CLI> 
[0KReliably Transmitting (no NAT) to 192.168.0.38:5651:
INVITE sip:2085@192.168.0.38:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK558e3487
Max-Forwards: 70
From: "89172011440" <sip:alpha-centre@192.168.2.90>;tag=as0871178a
To: <sip:2085@192.168.0.38:5651>
Contact: <sip:alpha-centre@192.168.2.90:5060>
Call-ID: 55a0fd1b1393cac66fdce475444b5d19@192.168.2.90:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.20.0
Date: Sun, 14 Jul 2013 07:11:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 258062721 258062721 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 19640 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.38:5651 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK66103e56
From: "89372542388"<sip:alpha-centre@192.168.2.90>;tag=as48533cf8
Call-ID: 1ab1c531084702b23ae602c83767f03e@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2085@192.168.0.38:5651>;tag=gncbngkp
Content-Length: 0
Warning: 399 ozsdk "User reject"

<------------->
--- (8 headers 0 lines) ---

[Ka01-centre*CLI> 
[0K    -- Got SIP response 486 "Busy Here" back from 192.168.0.38:5651

[Ka01-centre*CLI> 
[0KTransmitting (no NAT) to 192.168.0.38:5651:
ACK sip:2085@192.168.0.38:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK66103e56
Max-Forwards: 70
From: "89372542388" <sip:alpha-centre@192.168.2.90>;tag=as48533cf8
To: <sip:2085@192.168.0.38:5651>;tag=gncbngkp
Contact: <sip:alpha-centre@192.168.2.90:5060>
Call-ID: 1ab1c531084702b23ae602c83767f03e@192.168.2.90:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.20.0
Content-Length: 0


---

[Ka01-centre*CLI> 
[0K    -- SIP/2085-0000f947 is busy

[Ka01-centre*CLI> 
[0K    -- Nobody picked up in 2000 ms

[Ka01-centre*CLI> 
[0KReally destroying SIP dialog '1ab1c531084702b23ae602c83767f03e@192.168.2.90:5060' Method: INVITE

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.38:5651 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK558e3487
From: "89172011440"<sip:alpha-centre@192.168.2.90>;tag=as0871178a
Call-ID: 55a0fd1b1393cac66fdce475444b5d19@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2085@192.168.0.38:5651>
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.38:5651 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK66103e56
From: "89372542388"<sip:alpha-centre@192.168.2.90>;tag=as48533cf8
Call-ID: 1ab1c531084702b23ae602c83767f03e@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2085@192.168.0.38:5651>;tag=gncbngkp
Content-Length: 0
Warning: 399 ozsdk "User reject"

<------------->
--- (8 headers 0 lines) ---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.38:5651 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK558e3487
From: "89172011440"<sip:alpha-centre@192.168.2.90>;tag=as0871178a
Call-ID: 55a0fd1b1393cac66fdce475444b5d19@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2085@192.168.0.38:5651>;tag=gbjmindp
Content-Length: 0
Warning: 399 ozsdk "User reject"

<------------->
--- (8 headers 0 lines) ---
    -- Got SIP response 486 "Busy Here" back from 192.168.0.38:5651

[Ka01-centre*CLI> 
[0KTransmitting (no NAT) to 192.168.0.38:5651:
ACK sip:2085@192.168.0.38:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK558e3487
Max-Forwards: 70
From: "89172011440" <sip:alpha-centre@192.168.2.90>;tag=as0871178a
To: <sip:2085@192.168.0.38:5651>;tag=gbjmindp
Contact: <sip:alpha-centre@192.168.2.90:5060>
Call-ID: 55a0fd1b1393cac66fdce475444b5d19@192.168.2.90:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.20.0
Content-Length: 0


---

[Ka01-centre*CLI> 
[0K    -- SIP/2085-0000f949 is busy

[Ka01-centre*CLI> 
[0K    -- Nobody picked up in 1000 ms

[Ka01-centre*CLI> 
[0KReally destroying SIP dialog '55a0fd1b1393cac66fdce475444b5d19@192.168.2.90:5060' Method: INVITE

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.38:5651 --->
BYE sip:alpha-centre@192.168.2.90:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.38:5651;branch=z9hG4bK7b6f5e91-1947-4743-8cbc-8a42c1e79425;rport
To: "89873213690"<sip:alpha-centre@192.168.2.90>;tag=as1be3e29c
From: <sip:2085@192.168.0.38:5651>;tag=memutrwl
CSeq: 1 BYE
Call-ID: 581e47ea28cc941d7044d089575b49f5@192.168.2.90:5060
Max-Forwards: 70
Contact: <sip:2085@192.168.0.38:5651>
User-Agent: Ozeki VoIP SIP SDK
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

[Ka01-centre*CLI> 
[0KSending to 192.168.0.38:5651 (no NAT)

[Ka01-centre*CLI> 
[0KScheduling destruction of SIP dialog '581e47ea28cc941d7044d089575b49f5@192.168.2.90:5060' in 6400 ms (Method: BYE)

<--- Transmitting (no NAT) to 192.168.0.38:5651 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.38:5651;branch=z9hG4bK7b6f5e91-1947-4743-8cbc-8a42c1e79425;received=192.168.0.38;rport=5651
From: <sip:2085@192.168.0.38:5651>;tag=memutrwl
To: "89873213690"<sip:alpha-centre@192.168.2.90>;tag=as1be3e29c
Call-ID: 581e47ea28cc941d7044d089575b49f5@192.168.2.90:5060
CSeq: 1 BYE
Server: Asterisk PBX 1.8.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


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

[Ka01-centre*CLI> 
[0K  == Spawn extension (in_dial, 8452900000, 20) exited non-zero on 'SIP/alpha-centre-0000f935'
Scheduling destruction of SIP dialog '5b9e3dba75955e6e321fa57a266f1b0c@192.168.2.10' in 6400 ms (Method: ACK)

[Ka01-centre*CLI> 
[0Kset_destination: Parsing <sip:alpha-centre@192.168.2.10> for address/port to send to

[Ka01-centre*CLI> 
[0Kset_destination: set destination to 192.168.2.10:5060
Reliably Transmitting (no NAT) to 192.168.2.10:5060:
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


---

[Ka01-centre*CLI> 
[0K  == MixMonitor close filestream
  == End MixMonitor Recording SIP/alpha-centre-0000f935

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.2.10:5060 --->
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

<------------->
--- (10 headers 0 lines) ---

[Ka01-centre*CLI> 
[0KSIP Response message for INCOMING dialog BYE arrived

[Ka01-centre*CLI> 
[0KReally destroying SIP dialog '5b9e3dba75955e6e321fa57a266f1b0c@192.168.2.10' Method: ACK

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.2.60:5060 --->
OPTIONS sip:192.168.2.90 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.60:5060;branch=z9hG4bK195a2d5a;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.2.60>;tag=as15d9c085
To: <sip:192.168.2.90>
Contact: <sip:asterisk@192.168.2.60:5060>
Call-ID: 259c205b11b97ab477bb017f202a6e15@192.168.2.60:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.10.1
Date: Sun, 14 Jul 2013 07:11:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---

[Ka01-centre*CLI> 
[0KLooking for s in default (domain 192.168.2.90)

[Ka01-centre*CLI> 
[0K
<--- Transmitting (NAT) to 192.168.2.60:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.60:5060;branch=z9hG4bK195a2d5a;received=192.168.2.60;rport=5060
From: "asterisk" <sip:asterisk@192.168.2.60>;tag=as15d9c085
To: <sip:192.168.2.90>;tag=as79a0b151
Call-ID: 259c205b11b97ab477bb017f202a6e15@192.168.2.60:5060
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.8.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:192.168.2.90:5060>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '259c205b11b97ab477bb017f202a6e15@192.168.2.60:5060' in 32000 ms (Method: OPTIONS)

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.42:5651 --->
BYE sip:alpha-centre@192.168.2.90:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.42:5651;branch=z9hG4bK111d814d-ed33-4d73-b620-753a779e29f5;rport
To: "89616447484"<sip:alpha-centre@192.168.2.90>;tag=as257c1be1
From: <sip:2060@192.168.0.42:5651>;tag=jcnejwxu
CSeq: 1 BYE
Call-ID: 5092c5ec1b06ffa3194d08cc2a9e346d@192.168.2.90:5060
Max-Forwards: 70
Contact: <sip:2060@192.168.0.42:5651>
User-Agent: Ozeki VoIP SIP SDK
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

[Ka01-centre*CLI> 
[0KSending to 192.168.0.42:5651 (no NAT)

[Ka01-centre*CLI> 
[0KScheduling destruction of SIP dialog '5092c5ec1b06ffa3194d08cc2a9e346d@192.168.2.90:5060' in 6400 ms (Method: BYE)

[Ka01-centre*CLI> 
[0K
<--- Transmitting (no NAT) to 192.168.0.42:5651 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.42:5651;branch=z9hG4bK111d814d-ed33-4d73-b620-753a779e29f5;received=192.168.0.42;rport=5651
From: <sip:2060@192.168.0.42:5651>;tag=jcnejwxu
To: "89616447484"<sip:alpha-centre@192.168.2.90>;tag=as257c1be1
Call-ID: 5092c5ec1b06ffa3194d08cc2a9e346d@192.168.2.90:5060
CSeq: 1 BYE
Server: Asterisk PBX 1.8.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


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

[Ka01-centre*CLI> 
[0K  == Spawn extension (in_dial, 8452900000, 20) exited non-zero on 'SIP/alpha-centre-0000f938'

[Ka01-centre*CLI> 
[0KScheduling destruction of SIP dialog '4298227e1c2fcc8b0b287115245267e5@192.168.2.10' in 6400 ms (Method: ACK)

[Ka01-centre*CLI> 
[0Kset_destination: Parsing <sip:alpha-centre@192.168.2.10> for address/port to send to

[Ka01-centre*CLI> 
[0Kset_destination: set destination to 192.168.2.10:5060

[Ka01-centre*CLI> 
[0KReliably Transmitting (no NAT) to 192.168.2.10:5060:
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


---

[Ka01-centre*CLI> 
[0K  == MixMonitor close filestream

[Ka01-centre*CLI> 
[0K  == End MixMonitor Recording SIP/alpha-centre-0000f938

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.2.10:5060 --->
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

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '4298227e1c2fcc8b0b287115245267e5@192.168.2.10' Method: ACK

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.2.20:5060 --->
OPTIONS sip:192.168.2.90 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.20:5060;branch=z9hG4bK545b2b33
Max-Forwards: 70
From: "asterisk" <sip:beta-centre@192.168.2.20>;tag=as67b859c2
To: <sip:192.168.2.90>
Contact: <sip:beta-centre@192.168.2.20:5060>
Call-ID: 277e0c5b4af175fc163e492c7d18827a@192.168.2.20:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.10.1
Date: Sun, 14 Jul 2013 07:11:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---

[Ka01-centre*CLI> 
[0KLooking for s in default (domain 192.168.2.90)

<--- Transmitting (NAT) to 192.168.2.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.20:5060;branch=z9hG4bK545b2b33;received=192.168.2.20;rport=5060
From: "asterisk" <sip:beta-centre@192.168.2.20>;tag=as67b859c2
To: <sip:192.168.2.90>;tag=as1d21133a
Call-ID: 277e0c5b4af175fc163e492c7d18827a@192.168.2.20:5060
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.8.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:192.168.2.90:5060>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '277e0c5b4af175fc163e492c7d18827a@192.168.2.20:5060' in 32000 ms (Method: OPTIONS)

[Ka01-centre*CLI> 
[0K    -- Stopped music on hold on SIP/astr-centre-0000f93b

[Ka01-centre*CLI> 
[0K    -- <SIP/astr-centre-0000f93b> Playing 'queue-youarenext.gsm' (language 'ru')

[Ka01-centre*CLI> 
[0KReliably Transmitting (no NAT) to 192.168.0.36:5651:
OPTIONS sip:2038@192.168.0.36:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK17d20787
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.2.90>;tag=as3bc3fba1
To: <sip:2038@192.168.0.36:5651>
Contact: <sip:asterisk@192.168.2.90:5060>
Call-ID: 3b0ac6e227a56f9e5c95f23a4b18eb75@192.168.2.90:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.20.0
Date: Sun, 14 Jul 2013 07:11:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.36:5651 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK17d20787
From: "asterisk"<sip:asterisk@192.168.2.90>;tag=as3bc3fba1
Call-ID: 3b0ac6e227a56f9e5c95f23a4b18eb75@192.168.2.90:5060
CSeq: 102 OPTIONS
To: <sip:2038@192.168.0.36:5651>;tag=xmsqhilt
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

[Ka01-centre*CLI> 
[0KReally destroying SIP dialog '3b0ac6e227a56f9e5c95f23a4b18eb75@192.168.2.90:5060' Method: OPTIONS

[Ka01-centre*CLI> 
[0KReliably Transmitting (no NAT) to 192.168.2.20:5060:
OPTIONS sip:192.168.2.20 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK528b9cdd
Max-Forwards: 70
From: "asterisk" <sip:beta-centre@192.168.2.90>;tag=as035ab890
To: <sip:192.168.2.20>
Contact: <sip:beta-centre@192.168.2.90:5060>
Call-ID: 70bc0fb128c33a575e3d306f03c77226@192.168.2.90:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.20.0
Date: Sun, 14 Jul 2013 07:11:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.2.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK528b9cdd;received=192.168.2.90;rport=5060
From: "asterisk" <sip:beta-centre@192.168.2.90>;tag=as035ab890
To: <sip:192.168.2.20>;tag=as0da89fef
Call-ID: 70bc0fb128c33a575e3d306f03c77226@192.168.2.90:5060
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.8.10.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:192.168.2.20:5060>
Accept: application/sdp
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '70bc0fb128c33a575e3d306f03c77226@192.168.2.90:5060' Method: OPTIONS

[Ka01-centre*CLI> 
[0KReliably Transmitting (no NAT) to 192.168.0.38:5651:
OPTIONS sip:2085@192.168.0.38:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK75f51045
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.2.90>;tag=as61dbdb34
To: <sip:2085@192.168.0.38:5651>
Contact: <sip:asterisk@192.168.2.90:5060>
Call-ID: 260bc6c726004de7005921097561d5c6@192.168.2.90:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.20.0
Date: Sun, 14 Jul 2013 07:11:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.38:5651 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK75f51045
From: "asterisk"<sip:asterisk@192.168.2.90>;tag=as61dbdb34
Call-ID: 260bc6c726004de7005921097561d5c6@192.168.2.90:5060
CSeq: 102 OPTIONS
To: <sip:2085@192.168.0.38:5651>;tag=bbboivde
Content-Length: 0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Accept: application/sdp
Supported: 100rel
Accept-Language: en 

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

[Ka01-centre*CLI> 
[0K--- (11 headers 0 lines) ---
Really destroying SIP dialog '260bc6c726004de7005921097561d5c6@192.168.2.90:5060' Method: OPTIONS

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.2.10:5060 --->
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
<------------->
--- (14 headers 9 lines) ---

[Ka01-centre*CLI> 
[0KSending to 192.168.2.10:5060 (NAT)
Using INVITE request as basis request - 6ae8a8c42ab395e75280cbe56869f843@192.168.2.10
Found peer 'alpha-centre' for 'alpha-centre' from 192.168.2.10:5060
  == Using SIP RTP CoS mark 5
Found RTP audio format 8
Found audio description format PCMA for ID 8
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)

[Ka01-centre*CLI> 
[0KNon-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)

[Ka01-centre*CLI> 
[0KPeer audio RTP is at port 192.168.2.10:13416
Looking for 8452900000 in in_dial (domain 192.168.2.90)

[Ka01-centre*CLI> 
[0Klist_route: hop: <sip:alpha-centre@192.168.2.10>

<--- Transmitting (no NAT) to 192.168.2.10:5060 --->
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


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

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:1] [1;36mNoOp[0m("[1;35mSIP/alpha-centre-0000f94a[0m", "[1;35m<<<=== Входящий 89085570124 a01-centre-1373785915.63818 <<<===[0m") in new stack
    -- Executing [8452900000@in_dial:2] [1;36mSystem[0m("[1;35mSIP/alpha-centre-0000f94a[0m", "[1;35m/bin/echo "`date "+%F_%H-%M-%S"` "89085570124" <alpha-centre>" >> /tmp/calls-N.log[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:3] [1;36mGotoIf[0m("[1;35mSIP/alpha-centre-0000f94a[0m", "[1;35m0?To_Old[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:4] [1;36mGotoIf[0m("[1;35mSIP/alpha-centre-0000f94a[0m", "[1;35m0?blacklist[0m") in new stack
    -- Executing [8452900000@in_dial:5] [1;36mSet[0m("[1;35mSIP/alpha-centre-0000f94a[0m", "[1;35mOSTATE=0[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:6] [1;36mAGI[0m("[1;35mSIP/alpha-centre-0000f94a[0m", "[1;35mallo/get_order_state.php,89085570124[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Launched AGI Script /usr/share/asterisk/agi-bin/allo/get_order_state.php

[Ka01-centre*CLI> 
[0K[Jul 14 11:11:55] [0;31mERROR[0m[24240]: [1;37mutils.c[0m:[1;37m1169[0m [1;37mast_carefulwrite[0m: write() returned error: Broken pipe
    -- <SIP/alpha-centre-0000f94a>AGI Script allo/get_order_state.php completed, returning 0

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:7] [1;36mNoOp[0m("[1;35mSIP/alpha-centre-0000f94a[0m", "[1;35mOSTATE=10[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:8] [1;36mGotoIf[0m("[1;35mSIP/alpha-centre-0000f94a[0m", "[1;35m1?To_New[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Goto (in_dial,8452900000,15)
    -- Executing [8452900000@in_dial:15] [1;36mAnswer[0m("[1;35mSIP/alpha-centre-0000f94a[0m", "[1;35m[0m") in new stack

[Ka01-centre*CLI> 
[0KAudio is at 10492

[Ka01-centre*CLI> 
[0KAdding codec 0x8 (alaw) to SDP

[Ka01-centre*CLI> 
[0K
<--- Reliably Transmitting (no NAT) to 192.168.2.10:5060 --->
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

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

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.2.10:5060 --->
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

<------------->
--- (10 headers 0 lines) ---

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:16] [1;36mGosub[0m("[1;35mSIP/alpha-centre-0000f94a[0m", "[1;35mcall-record,,1[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@call-record:1] [1;36mGoto[0m("[1;35mSIP/alpha-centre-0000f94a[0m", "[1;35mrecord[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Goto (call-record,8452900000,5)

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@call-record:5] [1;36mMixMonitor[0m("[1;35mSIP/alpha-centre-0000f94a[0m", "[1;35m/var/spool/asterisk/monitor/record//2013/07/14/11_11_55_89085570124_8452900000_a01-centre-1373785915.63818.wav49,b[0m") in new stack

[Ka01-centre*CLI> 
[0K  == Begin MixMonitor Recording SIP/alpha-centre-0000f94a
    -- Executing [8452900000@call-record:6] [1;36mSet[0m("[1;35mSIP/alpha-centre-0000f94a[0m", "[1;35mCDR(userfield)=[0m") in new stack
    -- Executing [8452900000@call-record:7] [1;36mReturn[0m("[1;35mSIP/alpha-centre-0000f94a[0m", "[1;35m[0m") in new stack
    -- Executing [8452900000@in_dial:17] [1;36mGotoIf[0m("[1;35mSIP/alpha-centre-0000f94a[0m", "[1;35m0?setprior[0m") in new stack
    -- Executing [8452900000@in_dial:18] [1;36mSet[0m("[1;35mSIP/alpha-centre-0000f94a[0m", "[1;35mQUEUE_PRIO=5[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:19] [1;36mSystem[0m("[1;35mSIP/alpha-centre-0000f94a[0m", "[1;35m/bin/echo "`date "+%F_%H-%M-%S"` "89085570124" <alpha-centre>" >> /tmp/calls-NEW.log[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:20] [1;36mQueue[0m("[1;35mSIP/alpha-centre-0000f94a[0m", "[1;35mdispechers,tT,,,100[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Started music on hold, class 'default', on SIP/alpha-centre-0000f94a

[Ka01-centre*CLI> 
[0K  == Using SIP RTP CoS mark 5

[Ka01-centre*CLI> 
[0KAudio is at 13410
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

[Ka01-centre*CLI> 
[0KReliably Transmitting (no NAT) to 192.168.0.36:5651:
INVITE sip:2038@192.168.0.36:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK7498c8ff
Max-Forwards: 70
From: "89085570124" <sip:alpha-centre@192.168.2.90>;tag=as6bac7968
To: <sip:2038@192.168.0.36:5651>
Contact: <sip:alpha-centre@192.168.2.90:5060>
Call-ID: 1cdb262e14665e640d302d261f93f2e6@192.168.2.90:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.20.0
Date: Sun, 14 Jul 2013 07:11:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 1425324692 1425324692 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 13410 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.36:5651 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK7498c8ff
From: "89085570124"<sip:alpha-centre@192.168.2.90>;tag=as6bac7968
Call-ID: 1cdb262e14665e640d302d261f93f2e6@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2038@192.168.0.36:5651>
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.36:5651 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK7498c8ff
From: "89085570124"<sip:alpha-centre@192.168.2.90>;tag=as6bac7968
Call-ID: 1cdb262e14665e640d302d261f93f2e6@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2038@192.168.0.36:5651>;tag=jwdwhtlr
Content-Length: 0
Warning: 399 ozsdk "Busy"

<------------->
--- (8 headers 0 lines) ---

[Ka01-centre*CLI> 
[0K    -- Got SIP response 486 "Busy Here" back from 192.168.0.36:5651

[Ka01-centre*CLI> 
[0KTransmitting (no NAT) to 192.168.0.36:5651:
ACK sip:2038@192.168.0.36:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK7498c8ff
Max-Forwards: 70
From: "89085570124" <sip:alpha-centre@192.168.2.90>;tag=as6bac7968
To: <sip:2038@192.168.0.36:5651>;tag=jwdwhtlr
Contact: <sip:alpha-centre@192.168.2.90:5060>
Call-ID: 1cdb262e14665e640d302d261f93f2e6@192.168.2.90:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.20.0
Content-Length: 0


---

[Ka01-centre*CLI> 
[0K    -- SIP/2038-0000f94b is busy

[Ka01-centre*CLI> 
[0K    -- Nobody picked up in 0 ms

[Ka01-centre*CLI> 
[0K  == Using SIP RTP CoS mark 5

[Ka01-centre*CLI> 
[0KReally destroying SIP dialog '1cdb262e14665e640d302d261f93f2e6@192.168.2.90:5060' Method: INVITE

[Ka01-centre*CLI> 
[0KAudio is at 10426
Adding codec 0x2 (gsm) to SDP

[Ka01-centre*CLI> 
[0KAdding non-codec 0x1 (telephone-event) to SDP

[Ka01-centre*CLI> 
[0KReliably Transmitting (no NAT) to 192.168.0.42:5651:
INVITE sip:2060@192.168.0.42:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK3415baf3
Max-Forwards: 70
From: "89085570124" <sip:alpha-centre@192.168.2.90>;tag=as327397bd
To: <sip:2060@192.168.0.42:5651>
Contact: <sip:alpha-centre@192.168.2.90:5060>
Call-ID: 721d3c174e924fbc41918d5817b39c49@192.168.2.90:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.20.0
Date: Sun, 14 Jul 2013 07:11:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 1720468704 1720468704 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 10426 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.42:5651 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK3415baf3
From: "89085570124"<sip:alpha-centre@192.168.2.90>;tag=as327397bd
Call-ID: 721d3c174e924fbc41918d5817b39c49@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2060@192.168.0.42:5651>
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

[Ka01-centre*CLI> 
[0KReliably Transmitting (no NAT) to 192.168.2.60:5060:
OPTIONS sip:192.168.2.60 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK3c13647f
Max-Forwards: 70
From: "asterisk" <sip:epsilon@192.168.2.90>;tag=as26650084
To: <sip:192.168.2.60>
Contact: <sip:epsilon@192.168.2.90:5060>
Call-ID: 4909b13c02c2a6e36ed33736186ac4a2@192.168.2.90:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.20.0
Date: Sun, 14 Jul 2013 07:11:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.2.60:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK3c13647f;received=192.168.2.90;rport=5060
From: "asterisk" <sip:epsilon@192.168.2.90>;tag=as26650084
To: <sip:192.168.2.60>;tag=as651b8b0b
Call-ID: 4909b13c02c2a6e36ed33736186ac4a2@192.168.2.90:5060
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.8.10.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '4909b13c02c2a6e36ed33736186ac4a2@192.168.2.90:5060' Method: OPTIONS

[Ka01-centre*CLI> 
[0KReally destroying SIP dialog '581e47ea28cc941d7044d089575b49f5@192.168.2.90:5060' Method: BYE

[Ka01-centre*CLI> 
[0KReally destroying SIP dialog '68d8724730df90e150c694ef5594a036@192.168.2.10' Method: OPTIONS

[Ka01-centre*CLI> 
[0KReliably Transmitting (no NAT) to 192.168.0.39:5651:
OPTIONS sip:2068@192.168.0.39:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK6087f7c4
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.2.90>;tag=as47f70deb
To: <sip:2068@192.168.0.39:5651>
Contact: <sip:asterisk@192.168.2.90:5060>
Call-ID: 1d1601b03ce4115b1063efc962e6223d@192.168.2.90:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.20.0
Date: Sun, 14 Jul 2013 07:11:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.39:5651 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK6087f7c4
From: "asterisk"<sip:asterisk@192.168.2.90>;tag=as47f70deb
Call-ID: 1d1601b03ce4115b1063efc962e6223d@192.168.2.90:5060
CSeq: 102 OPTIONS
To: <sip:2068@192.168.0.39:5651>;tag=mmimlmoi
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '1d1601b03ce4115b1063efc962e6223d@192.168.2.90:5060' Method: OPTIONS

[Ka01-centre*CLI> 
[0K    -- Stopped music on hold on SIP/alpha-centre-0000f940

[Ka01-centre*CLI> 
[0K    -- <SIP/alpha-centre-0000f940> Playing 'queue-thereare.alaw' (language 'ru')

[Ka01-centre*CLI> 
[0K    -- Told SIP/astr-centre-0000f93b in dispechers their queue position (which was 1)

[Ka01-centre*CLI> 
[0K    -- <SIP/astr-centre-0000f93b> Playing 'queue-thankyou.gsm' (language 'ru')

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.42:5651 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK3415baf3
From: "89085570124"<sip:alpha-centre@192.168.2.90>;tag=as327397bd
Call-ID: 721d3c174e924fbc41918d5817b39c49@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2060@192.168.0.42:5651>;tag=xmlranki
Content-Type: application/sdp
Content-Length: 200
Contact: <sip:2060@192.168.0.42:5651>

v=0
o=- 2044424519 2044424519 IN IP4 192.168.0.42
s=Ozeki VoIP SIP SDK
c=IN IP4 192.168.0.42
t=0 0
m=audio 5659 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
--- (9 headers 9 lines) ---

[Ka01-centre*CLI> 
[0KFound RTP audio format 3
Found RTP audio format 101
Found audio description format GSM for ID 3

[Ka01-centre*CLI> 
[0KFound audio description format telephone-event for ID 101
Capabilities: us - 0x2 (gsm), peer - audio=0x2 (gsm)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x2 (gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.42:5659

[Ka01-centre*CLI> 
[0Klist_route: hop: <sip:2060@192.168.0.42:5651>

[Ka01-centre*CLI> 
[0Kset_destination: Parsing <sip:2060@192.168.0.42:5651> for address/port to send to
set_destination: set destination to 192.168.0.42:5651

[Ka01-centre*CLI> 
[0KTransmitting (no NAT) to 192.168.0.42:5651:
ACK sip:2060@192.168.0.42:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK505c4aba
Max-Forwards: 70
From: "89085570124" <sip:alpha-centre@192.168.2.90>;tag=as327397bd
To: <sip:2060@192.168.0.42:5651>;tag=xmlranki
Contact: <sip:alpha-centre@192.168.2.90:5060>
Call-ID: 721d3c174e924fbc41918d5817b39c49@192.168.2.90:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.20.0
Content-Length: 0


---

[Ka01-centre*CLI> 
[0K    -- SIP/2060-0000f94c answered SIP/alpha-centre-0000f94a

[Ka01-centre*CLI> 
[0K    -- Stopped music on hold on SIP/alpha-centre-0000f94a

[Ka01-centre*CLI> 
[0KReally destroying SIP dialog '5092c5ec1b06ffa3194d08cc2a9e346d@192.168.2.90:5060' Method: BYE

[Ka01-centre*CLI> 
[0K    -- Stopped music on hold on SIP/alpha-centre-0000f943

[Ka01-centre*CLI> 
[0K    -- <SIP/alpha-centre-0000f943> Playing 'queue-thereare.alaw' (language 'ru')

[Ka01-centre*CLI> 
[0K    -- Started music on hold, class 'default', on SIP/astr-centre-0000f93b

[Ka01-centre*CLI> 
[0K  == Using SIP RTP CoS mark 5

[Ka01-centre*CLI> 
[0KAudio is at 18242
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.0.39:5651:
INVITE sip:2068@192.168.0.39:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK26a70880
Max-Forwards: 70
From: "88452445099" <sip:astr-centre@192.168.2.90>;tag=as0b94e2aa
To: <sip:2068@192.168.0.39:5651>
Contact: <sip:astr-centre@192.168.2.90:5060>
Call-ID: 74a383a5028d52982788d016089a26f4@192.168.2.90:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.20.0
Date: Sun, 14 Jul 2013 07:11:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 1445520743 1445520743 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 18242 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.39:5651 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK26a70880
From: "88452445099"<sip:astr-centre@192.168.2.90>;tag=as0b94e2aa
Call-ID: 74a383a5028d52982788d016089a26f4@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2068@192.168.0.39:5651>
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.39:5651 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK26a70880
From: "88452445099"<sip:astr-centre@192.168.2.90>;tag=as0b94e2aa
Call-ID: 74a383a5028d52982788d016089a26f4@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2068@192.168.0.39:5651>;tag=moujkndh
Content-Length: 0
Warning: 399 ozsdk "Busy"

<------------->
--- (8 headers 0 lines) ---
    -- Got SIP response 486 "Busy Here" back from 192.168.0.39:5651
Transmitting (no NAT) to 192.168.0.39:5651:
ACK sip:2068@192.168.0.39:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK26a70880
Max-Forwards: 70
From: "88452445099" <sip:astr-centre@192.168.2.90>;tag=as0b94e2aa
To: <sip:2068@192.168.0.39:5651>;tag=moujkndh
Contact: <sip:astr-centre@192.168.2.90:5060>
Call-ID: 74a383a5028d52982788d016089a26f4@192.168.2.90:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.20.0
Content-Length: 0


---

[Ka01-centre*CLI> 
[0K    -- SIP/2068-0000f94d is busy

[Ka01-centre*CLI> 
[0K    -- Nobody picked up in 0 ms

[Ka01-centre*CLI> 
[0K  == Using SIP RTP CoS mark 5

[Ka01-centre*CLI> 
[0KReally destroying SIP dialog '74a383a5028d52982788d016089a26f4@192.168.2.90:5060' Method: INVITE
Audio is at 16892
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

[Ka01-centre*CLI> 
[0KReliably Transmitting (no NAT) to 192.168.0.38:5651:
INVITE sip:2085@192.168.0.38:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK3c39782c
Max-Forwards: 70
From: "88452445099" <sip:astr-centre@192.168.2.90>;tag=as3cb60605
To: <sip:2085@192.168.0.38:5651>
Contact: <sip:astr-centre@192.168.2.90:5060>
Call-ID: 5d205f0059d5dd8947664494075cd761@192.168.2.90:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.20.0
Date: Sun, 14 Jul 2013 07:11:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 1583854896 1583854896 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 16892 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.38:5651 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK3c39782c
From: "88452445099"<sip:astr-centre@192.168.2.90>;tag=as3cb60605
Call-ID: 5d205f0059d5dd8947664494075cd761@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2085@192.168.0.38:5651>
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

[Ka01-centre*CLI> 
[0K    -- <SIP/alpha-centre-0000f940> Playing 'digits/2.alaw' (language 'ru')

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.2.10:5060 --->
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
<------------->
--- (14 headers 9 lines) ---

[Ka01-centre*CLI> 
[0KSending to 192.168.2.10:5060 (NAT)
Using INVITE request as basis request - 447ff969744d92fb4653955866a164ed@192.168.2.10

[Ka01-centre*CLI> 
[0KFound peer 'alpha-centre' for 'alpha-centre' from 192.168.2.10:5060

[Ka01-centre*CLI> 
[0K  == Using SIP RTP CoS mark 5

[Ka01-centre*CLI> 
[0KFound RTP audio format 8
Found audio description format PCMA for ID 8
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.2.10:15676

[Ka01-centre*CLI> 
[0KLooking for 8452900000 in in_dial (domain 192.168.2.90)

[Ka01-centre*CLI> 
[0Klist_route: hop: <sip:alpha-centre@192.168.2.10>

[Ka01-centre*CLI> 
[0K
<--- Transmitting (no NAT) to 192.168.2.10:5060 --->
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


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

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:1] [1;36mNoOp[0m("[1;35mSIP/alpha-centre-0000f94f[0m", "[1;35m<<<=== Входящий 89276220039 a01-centre-1373785919.63823 <<<===[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:2] [1;36mSystem[0m("[1;35mSIP/alpha-centre-0000f94f[0m", "[1;35m/bin/echo "`date "+%F_%H-%M-%S"` "89276220039" <alpha-centre>" >> /tmp/calls-N.log[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:3] [1;36mGotoIf[0m("[1;35mSIP/alpha-centre-0000f94f[0m", "[1;35m0?To_Old[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:4] [1;36mGotoIf[0m("[1;35mSIP/alpha-centre-0000f94f[0m", "[1;35m0?blacklist[0m") in new stack
    -- Executing [8452900000@in_dial:5] [1;36mSet[0m("[1;35mSIP/alpha-centre-0000f94f[0m", "[1;35mOSTATE=0[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:6] [1;36mAGI[0m("[1;35mSIP/alpha-centre-0000f94f[0m", "[1;35mallo/get_order_state.php,89276220039[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Launched AGI Script /usr/share/asterisk/agi-bin/allo/get_order_state.php

[Ka01-centre*CLI> 
[0K    -- <SIP/alpha-centre-0000f940> Playing 'queue-callswaiting.alaw' (language 'ru')

[Ka01-centre*CLI> 
[0K[Jul 14 11:12:00] [0;31mERROR[0m[24254]: [1;37mutils.c[0m:[1;37m1169[0m [1;37mast_carefulwrite[0m: write() returned error: Broken pipe
    -- <SIP/alpha-centre-0000f94f>AGI Script allo/get_order_state.php completed, returning 0

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:7] [1;36mNoOp[0m("[1;35mSIP/alpha-centre-0000f94f[0m", "[1;35mOSTATE=10[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:8] [1;36mGotoIf[0m("[1;35mSIP/alpha-centre-0000f94f[0m", "[1;35m1?To_New[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Goto (in_dial,8452900000,15)

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:15] [1;36mAnswer[0m("[1;35mSIP/alpha-centre-0000f94f[0m", "[1;35m[0m") in new stack

[Ka01-centre*CLI> 
[0KAudio is at 11470
Adding codec 0x8 (alaw) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.2.10:5060 --->
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

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

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.2.10:5060 --->
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

<------------->
--- (10 headers 0 lines) ---

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:16] [1;36mGosub[0m("[1;35mSIP/alpha-centre-0000f94f[0m", "[1;35mcall-record,,1[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@call-record:1] [1;36mGoto[0m("[1;35mSIP/alpha-centre-0000f94f[0m", "[1;35mrecord[0m") in new stack
    -- Goto (call-record,8452900000,5)
    -- Executing [8452900000@call-record:5] [1;36mMixMonitor[0m("[1;35mSIP/alpha-centre-0000f94f[0m", "[1;35m/var/spool/asterisk/monitor/record//2013/07/14/11_12_00_89276220039_8452900000_a01-centre-1373785919.63823.wav49,b[0m") in new stack

[Ka01-centre*CLI> 
[0K  == Begin MixMonitor Recording SIP/alpha-centre-0000f94f
    -- Executing [8452900000@call-record:6] [1;36mSet[0m("[1;35mSIP/alpha-centre-0000f94f[0m", "[1;35mCDR(userfield)=[0m") in new stack
    -- Executing [8452900000@call-record:7] [1;36mReturn[0m("[1;35mSIP/alpha-centre-0000f94f[0m", "[1;35m[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:17] [1;36mGotoIf[0m("[1;35mSIP/alpha-centre-0000f94f[0m", "[1;35m0?setprior[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:18] [1;36mSet[0m("[1;35mSIP/alpha-centre-0000f94f[0m", "[1;35mQUEUE_PRIO=5[0m") in new stack
    -- Executing [8452900000@in_dial:19] [1;36mSystem[0m("[1;35mSIP/alpha-centre-0000f94f[0m", "[1;35m/bin/echo "`date "+%F_%H-%M-%S"` "89276220039" <alpha-centre>" >> /tmp/calls-NEW.log[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Executing [8452900000@in_dial:20] [1;36mQueue[0m("[1;35mSIP/alpha-centre-0000f94f[0m", "[1;35mdispechers,tT,,,100[0m") in new stack

[Ka01-centre*CLI> 
[0K    -- Started music on hold, class 'default', on SIP/alpha-centre-0000f94f

[Ka01-centre*CLI> 
[0K  == Using SIP RTP CoS mark 5

[Ka01-centre*CLI> 
[0KAudio is at 15598
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

[Ka01-centre*CLI> 
[0KReliably Transmitting (no NAT) to 192.168.0.39:5651:
INVITE sip:2068@192.168.0.39:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK77951a7e
Max-Forwards: 70
From: "89276220039" <sip:alpha-centre@192.168.2.90>;tag=as1adaca46
To: <sip:2068@192.168.0.39:5651>
Contact: <sip:alpha-centre@192.168.2.90:5060>
Call-ID: 11e03ef61ca05a736bf8c18c34c90d5e@192.168.2.90:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.20.0
Date: Sun, 14 Jul 2013 07:12:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 1003148663 1003148663 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 15598 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.39:5651 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK77951a7e
From: "89276220039"<sip:alpha-centre@192.168.2.90>;tag=as1adaca46
Call-ID: 11e03ef61ca05a736bf8c18c34c90d5e@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2068@192.168.0.39:5651>
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.39:5651 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK77951a7e
From: "89276220039"<sip:alpha-centre@192.168.2.90>;tag=as1adaca46
Call-ID: 11e03ef61ca05a736bf8c18c34c90d5e@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2068@192.168.0.39:5651>;tag=synywish
Content-Length: 0
Warning: 399 ozsdk "Busy"

<------------->
--- (8 headers 0 lines) ---
    -- Got SIP response 486 "Busy Here" back from 192.168.0.39:5651

[Ka01-centre*CLI> 
[0KTransmitting (no NAT) to 192.168.0.39:5651:
ACK sip:2068@192.168.0.39:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK77951a7e
Max-Forwards: 70
From: "89276220039" <sip:alpha-centre@192.168.2.90>;tag=as1adaca46
To: <sip:2068@192.168.0.39:5651>;tag=synywish
Contact: <sip:alpha-centre@192.168.2.90:5060>
Call-ID: 11e03ef61ca05a736bf8c18c34c90d5e@192.168.2.90:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.20.0
Content-Length: 0


---

[Ka01-centre*CLI> 
[0K    -- SIP/2068-0000f950 is busy

[Ka01-centre*CLI> 
[0K    -- Nobody picked up in 0 ms

[Ka01-centre*CLI> 
[0K  == Using SIP RTP CoS mark 5

[Ka01-centre*CLI> 
[0KReally destroying SIP dialog '11e03ef61ca05a736bf8c18c34c90d5e@192.168.2.90:5060' Method: INVITE

[Ka01-centre*CLI> 
[0KAudio is at 12308
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

[Ka01-centre*CLI> 
[0KReliably Transmitting (no NAT) to 192.168.0.38:5651:
INVITE sip:2085@192.168.0.38:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK1105c87b
Max-Forwards: 70
From: "89276220039" <sip:alpha-centre@192.168.2.90>;tag=as497abc40
To: <sip:2085@192.168.0.38:5651>
Contact: <sip:alpha-centre@192.168.2.90:5060>
Call-ID: 0ca0899f32d90f5d5f63fef70fdcef01@192.168.2.90:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.20.0
Date: Sun, 14 Jul 2013 07:12:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 529302333 529302333 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 12308 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

[Ka01-centre*CLI> 
[0KRetransmitting #1 (no NAT) to 192.168.0.38:5651:
INVITE sip:2085@192.168.0.38:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK1105c87b
Max-Forwards: 70
From: "89276220039" <sip:alpha-centre@192.168.2.90>;tag=as497abc40
To: <sip:2085@192.168.0.38:5651>
Contact: <sip:alpha-centre@192.168.2.90:5060>
Call-ID: 0ca0899f32d90f5d5f63fef70fdcef01@192.168.2.90:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.20.0
Date: Sun, 14 Jul 2013 07:12:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 529302333 529302333 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 12308 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

[Ka01-centre*CLI> 
[0KRetransmitting #2 (no NAT) to 192.168.0.38:5651:
INVITE sip:2085@192.168.0.38:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK1105c87b
Max-Forwards: 70
From: "89276220039" <sip:alpha-centre@192.168.2.90>;tag=as497abc40
To: <sip:2085@192.168.0.38:5651>
Contact: <sip:alpha-centre@192.168.2.90:5060>
Call-ID: 0ca0899f32d90f5d5f63fef70fdcef01@192.168.2.90:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.20.0
Date: Sun, 14 Jul 2013 07:12:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 529302333 529302333 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 12308 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

[Ka01-centre*CLI> 
[0KRetransmitting #3 (no NAT) to 192.168.0.38:5651:
INVITE sip:2085@192.168.0.38:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK1105c87b
Max-Forwards: 70
From: "89276220039" <sip:alpha-centre@192.168.2.90>;tag=as497abc40
To: <sip:2085@192.168.0.38:5651>
Contact: <sip:alpha-centre@192.168.2.90:5060>
Call-ID: 0ca0899f32d90f5d5f63fef70fdcef01@192.168.2.90:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.20.0
Date: Sun, 14 Jul 2013 07:12:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 529302333 529302333 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 12308 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

[Ka01-centre*CLI> 
[0K    -- <SIP/alpha-centre-0000f943> Playing 'digits/3.alaw' (language 'ru')

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.38:5651 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK1105c87b
From: "89276220039"<sip:alpha-centre@192.168.2.90>;tag=as497abc40
Call-ID: 0ca0899f32d90f5d5f63fef70fdcef01@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2085@192.168.0.38:5651>
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.38:5651 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK3c39782c
From: "88452445099"<sip:astr-centre@192.168.2.90>;tag=as3cb60605
Call-ID: 5d205f0059d5dd8947664494075cd761@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2085@192.168.0.38:5651>;tag=jornhiap
Content-Type: application/sdp
Content-Length: 198
Contact: <sip:2085@192.168.0.38:5651>

v=0
o=- 784959801 784959801 IN IP4 192.168.0.38
s=Ozeki VoIP SIP SDK
c=IN IP4 192.168.0.38
t=0 0
m=audio 5737 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
--- (9 headers 9 lines) ---

[Ka01-centre*CLI> 
[0KFound RTP audio format 3
Found RTP audio format 101
Found audio description format GSM for ID 3

[Ka01-centre*CLI> 
[0KFound audio description format telephone-event for ID 101
Capabilities: us - 0x2 (gsm), peer - audio=0x2 (gsm)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x2 (gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.38:5737
list_route: hop: <sip:2085@192.168.0.38:5651>

[Ka01-centre*CLI> 
[0Kset_destination: Parsing <sip:2085@192.168.0.38:5651> for address/port to send to

[Ka01-centre*CLI> 
[0Kset_destination: set destination to 192.168.0.38:5651
Transmitting (no NAT) to 192.168.0.38:5651:
ACK sip:2085@192.168.0.38:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK44689169
Max-Forwards: 70
From: "88452445099" <sip:astr-centre@192.168.2.90>;tag=as3cb60605
To: <sip:2085@192.168.0.38:5651>;tag=jornhiap
Contact: <sip:astr-centre@192.168.2.90:5060>
Call-ID: 5d205f0059d5dd8947664494075cd761@192.168.2.90:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.20.0
Content-Length: 0


-
[Ka01-centre*CLI> 
[0K--

[Ka01-centre*CLI> 
[0K    -- SIP/2085-0000f94e answered SIP/astr-centre-0000f93b

[Ka01-centre*CLI> 
[0K    -- Stopped music on hold on SIP/astr-centre-0000f93b

[Ka01-centre*CLI> 
[0K    -- <SIP/alpha-centre-0000f943> Playing 'queue-callswaiting.alaw' (language 'ru')

[Ka01-centre*CLI> 
[0K
<--- SIP read from UDP:192.168.0.38:5651 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK3c39782c
From: "88452445099"<sip:astr-centre@192.168.2.90>;tag=as3cb60605
Call-ID: 5d205f0059d5dd8947664494075cd761@192.168.2.90:5060
CSeq: 102 INVITE
To: <sip:2085@192.168.0.38:5651>;tag=jornhiap
Content-Type: application/sdp
Content-Length: 198
Contact: <sip:2085@192.168.0.38:5651>

v=0
o=- 784959801 784959801 IN IP4 192.168.0.38
s=Ozeki VoIP SIP SDK
c=IN IP4 192.168.0.38
t=0 0
m=audio 5737 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
--- (9 headers 9 lines) ---
set_destination: Parsing <sip:2085@192.168.0.38:5651> for address/port to send to

[Ka01-centre*CLI> 
[0Kset_destination: set destination to 192.168.0.38:5651

[Ka01-centre*CLI> 
[0KTransmitting (no NAT) to 192.168.0.38:5651:
ACK sip:2085@192.168.0.38:5651 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.90:5060;branch=z9hG4bK5de1f3b4
Max-Forwards: 70
From: "88452445099" <sip:astr-centre@192.168.2.90>;tag=as3cb60605
To: <sip:2085@192.168.0.38:5651>;tag=jornhiap
Contact: <sip:astr-centre@192.168.2.90:5060>
Call-ID: 5d205f0059d5dd8947664494075cd761@192.168.2.90:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.20.0
Content-Length: 0


---

[Ka01-centre*CLI> 
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups
Asterisk ending (0).
