
v=0
o=BroadWorks 1225093262 1 IN IP4 92.46.61.21
s=-
c=IN IP4 92.46.61.21
t=0 0
m=audio 14012 RTP/AVP 8 18 101
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

<------------->
--- (13 headers 10 lines) ---
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP TOS bits 136
  == Using SIP VRTP CoS mark 6
Sending to 92.46.61.21 : 5060 (no NAT)
Using INVITE request as basis request - BW133932991110411-948823542@10.14.0.2
No user '87272445474' in SIP users list
Found peer 'telecom-3907000' for '87272445474' from 92.46.61.21:5060
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - 0x28000c (ulaw|alaw|h263|h264), peer - audio=0x108 (alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 92.46.61.21:14012
Peer doesn't provide video
Looking for 3907000 in from-trunk (domain 192.168.1.10)
list_route: hop: <sip:87272445474@92.46.61.21:5060;transport=udp>
trixbox1*CLI>
<--- Transmitting (no NAT) to 92.46.61.21:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 92.46.61.21:5060;branch=z9hG4bK6l1j9f1028k0cq0e8180.1;received=92.46.61.21
From: <sip:87272445474@10.14.0.2;user=phone>;tag=355664481-1302507572991-
To: "John BABII GALINA VASILEVNA"<sip:133930188999036@sip.telecom.kz;ep=95.56.104.24:5060;fw=95.56.104.24:5060>
Call-ID: BW133932991110411-948823542@10.14.0.2
CSeq: 566241152 INVITE
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:3907000@192.168.1.10>
Content-Length: 0


<------------>
    -- Executing [3907000@from-trunk:1] Set("SIP/telecom-3907000-0000000c", "__FROM_DID=3907000") in new stack
    -- Executing [3907000@from-trunk:2] Gosub("SIP/telecom-3907000-0000000c", "app-blacklist-check,s,1") in new stack
    -- Executing [s@app-blacklist-check:1] GotoIf("SIP/telecom-3907000-0000000c", "0?blacklisted") in new stack
    -- Executing [s@app-blacklist-check:2] Return("SIP/telecom-3907000-0000000c", "") in new stack
    -- Executing [3907000@from-trunk:3] ExecIf("SIP/telecom-3907000-0000000c", "1 ?Set(CALLERID(name)=87272445474)") in new stack
    -- Executing [3907000@from-trunk:4] Set("SIP/telecom-3907000-0000000c", "FAX_RX=system") in new stack
    -- Executing [3907000@from-trunk:5] Set("SIP/telecom-3907000-0000000c", "FAX_RX_EMAIL=aslan.tumenov@gmail.com") in new stack
    -- Executing [3907000@from-trunk:6] Answer("SIP/telecom-3907000-0000000c", "") in new stack
Audio is at 192.168.1.10 port 44382
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
trixbox1*CLI>
<--- Reliably Transmitting (no NAT) to 92.46.61.21:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 92.46.61.21:5060;branch=z9hG4bK6l1j9f1028k0cq0e8180.1;received=92.46.61.21
From: <sip:87272445474@10.14.0.2;user=phone>;tag=355664481-1302507572991-
To: "John BABII GALINA VASILEVNA"<sip:133930188999036@sip.telecom.kz;ep=95.56.104.24:5060;fw=95.56.104.24:5060>;tag=as79d0413f
Call-ID: BW133932991110411-948823542@10.14.0.2
CSeq: 566241152 INVITE
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:3907000@192.168.1.10>
Content-Type: application/sdp
Content-Length: 276

v=0
o=root 1387193981 1387193981 IN IP4 192.168.1.10
s=Asterisk PBX 1.6.0.26-FONCORE-r78
c=IN IP4 192.168.1.10
t=0 0
m=audio 44382 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
trixbox1*CLI>
<--- SIP read from UDP://92.46.61.21:5060 --->
ACK sip:3907000@192.168.1.10:5060 SIP/2.0
Via: SIP/2.0/UDP 92.46.61.21:5060;branch=z9hG4bKadf18e3048ggdq0et0s0.1
From: <sip:87272445474@10.14.0.2;user=phone>;tag=355664481-1302507572991-
To: "John BABII GALINA VASILEVNA"<sip:133930188999036@sip.telecom.kz;ep=95.56.104.24:5060;fw=95.56.104.24:5060>;tag=as79d0413f
Call-ID: BW133932991110411-948823542@10.14.0.2
CSeq: 566241152 ACK
Contact: <sip:87272445474@92.46.61.21:5060;transport=udp>
Max-Forwards: 69
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
    -- Executing [3907000@from-trunk:7] PlayTones("SIP/telecom-3907000-0000000c", "ring") in new stack
    -- Executing [3907000@from-trunk:8] NVFaxDetect("SIP/telecom-3907000-0000000c", "4|t") in new stack
    -- Executing [3907000@from-trunk:9] Set("SIP/telecom-3907000-0000000c", "__CALLINGPRES_SV=allowed_not_screened") in new stack
    -- Executing [3907000@from-trunk:10] Set("SIP/telecom-3907000-0000000c", "CALLERPRES()=allowed_not_screened") in new stack
    -- Executing [3907000@from-trunk:11] Goto("SIP/telecom-3907000-0000000c", "from-did-direct,102,1") in new stack
    -- Goto (from-did-direct,102,1)
    -- Executing [102@from-did-direct:1] Macro("SIP/telecom-3907000-0000000c", "exten-vm,novm,102") in new stack
    -- Executing [s@macro-exten-vm:1] Macro("SIP/telecom-3907000-0000000c", "user-callerid") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/telecom-3907000-0000000c", "AMPUSER=87272445474") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/telecom-3907000-0000000c", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/telecom-3907000-0000000c", "1?Set(REALCALLERIDNUM=87272445474)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/telecom-3907000-0000000c", "AMPUSER=") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/telecom-3907000-0000000c", "AMPUSERCIDNAME=") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/telecom-3907000-0000000c", "1?report") in new stack
    -- Goto (macro-user-callerid,s,10)
    -- Executing [s@macro-user-callerid:10] GotoIf("SIP/telecom-3907000-0000000c", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:11] Set("SIP/telecom-3907000-0000000c", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:12] GotoIf("SIP/telecom-3907000-0000000c", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [s@macro-user-callerid:19] NoOp("SIP/telecom-3907000-0000000c", "Using CallerID "87272445474" <87272445474>") in new stack
    -- Executing [s@macro-exten-vm:2] Set("SIP/telecom-3907000-0000000c", "RingGroupMethod=none") in new stack
    -- Executing [s@macro-exten-vm:3] Set("SIP/telecom-3907000-0000000c", "VMBOX=novm") in new stack
    -- Executing [s@macro-exten-vm:4] Set("SIP/telecom-3907000-0000000c", "EXTTOCALL=102") in new stack
    -- Executing [s@macro-exten-vm:5] Set("SIP/telecom-3907000-0000000c", "CFUEXT=") in new stack
    -- Executing [s@macro-exten-vm:6] Set("SIP/telecom-3907000-0000000c", "CFBEXT=") in new stack
    -- Executing [s@macro-exten-vm:7] Set("SIP/telecom-3907000-0000000c", "RT=""") in new stack
    -- Executing [s@macro-exten-vm:8] Macro("SIP/telecom-3907000-0000000c", "record-enable,102,IN") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/telecom-3907000-0000000c", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] AGI("SIP/telecom-3907000-0000000c", "recordingcheck,20110411-133938,1302507573.12") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
 recordingcheck,20110411-133938,1302507573.12: Inbound recording not enabled
    -- <SIP/telecom-3907000-0000000c>AGI Script recordingcheck completed, returning 0
    -- Executing [s@macro-record-enable:5] MacroExit("SIP/telecom-3907000-0000000c", "") in new stack
    -- Executing [s@macro-exten-vm:9] Macro("SIP/telecom-3907000-0000000c", "dial,"",tr,102") in new stack
    -- Executing [s@macro-dial:1] GotoIf("SIP/telecom-3907000-0000000c", "1?dial") in new stack
    -- Goto (macro-dial,s,3)
    -- Executing [s@macro-dial:3] AGI("SIP/telecom-3907000-0000000c", "dialparties.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
 dialparties.agi: Starting New Dialparties.agi
 dialparties.agi: Caller ID name is '87272445474' number is '87272445474'
 dialparties.agi: Methodology of ring is  'none'
    -- dialparties.agi: Added extension 102 to extension map
    -- dialparties.agi: Extension 102 cf is disabled
    -- dialparties.agi: Extension 102 do not disturb is disabled
 dialparties.agi: EXTENSION_STATE: 1 (INUSE)
 dialparties.agi: Extension 102 has call waiting enabled with state: 1
    -- dialparties.agi: dbset CALLTRACE/102 to 87272445474
    -- dialparties.agi: Filtered ARG3: 102
    -- <SIP/telecom-3907000-0000000c>AGI Script dialparties.agi completed, returning 0
    -- Executing [s@macro-dial:7] Dial("SIP/telecom-3907000-0000000c", "SIP/102,"",tr") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP TOS bits 136
  == Using SIP VRTP CoS mark 6
    -- Called 102
    -- SIP/102-0000000d is ringing
    -- Started music on hold, class 'default', on SIP/telecom-3907000-0000000a
    -- SIP/102-0000000d answered SIP/telecom-3907000-0000000c
Really destroying SIP dialog '44837ac22031494f6fd6c24606fd457d@127.0.0.1' Method: REGISTER
    -- Executing [h@macro-dial:1] Macro("SIP/telecom-3907000-0000000c", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/telecom-3907000-0000000c", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,4)
    -- Executing [s@macro-hangupcall:4] GotoIf("SIP/telecom-3907000-0000000c", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/telecom-3907000-0000000c", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] Hangup("SIP/telecom-3907000-0000000c", "") in new stack
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/telecom-3907000-0000000c' in macro 'hangupcall'
  == Spawn extension (macro-dial, h, 1) exited non-zero on 'SIP/telecom-3907000-0000000c'
  == Spawn extension (macro-dial, s, 7) exited non-zero on 'SIP/telecom-3907000-0000000c' in macro 'dial'
  == Spawn extension (macro-exten-vm, s, 9) exited non-zero on 'SIP/telecom-3907000-0000000c' in macro 'exten-vm'
  == Spawn extension (from-did-direct, 102, 1) exited non-zero on 'SIP/telecom-3907000-0000000c'
Scheduling destruction of SIP dialog 'BW133932991110411-948823542@10.14.0.2' in 6400 ms (Method: ACK)
set_destination: Parsing <sip:87272445474@92.46.61.21:5060;transport=udp> for address/port to send to
set_destination: set destination to 92.46.61.21, port 5060
Reliably Transmitting (no NAT) to 92.46.61.21:5060:
BYE sip:87272445474@92.46.61.21:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK03b936dd;rport
Max-Forwards: 70
From: "John BABII GALINA VASILEVNA"<sip:133930188999036@sip.telecom.kz;ep=95.56.104.24:5060;fw=95.56.104.24:5060>;tag=as79d0413f
To: <sip:87272445474@10.14.0.2;user=phone>;tag=355664481-1302507572991-
Call-ID: BW133932991110411-948823542@10.14.0.2
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
trixbox1*CLI>
<--- SIP read from UDP://92.46.61.21:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.10:5060;received=95.56.104.24;branch=z9hG4bK03b936dd;rport=5060
From: "John BABII GALINA VASILEVNA"<sip:133930188999036@sip.telecom.kz;ep=95.56.104.24:5060;fw=95.56.104.24:5060>;tag=as79d0413f
To: <sip:87272445474@10.14.0.2;user=phone>;tag=355664481-1302507572991-
Call-ID: BW133932991110411-948823542@10.14.0.2
CSeq: 102 BYE
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog 'BW133932991110411-948823542@10.14.0.2' Method: ACK
trixbox1*CLI>
<--- SIP read from UDP://92.46.61.21:5060 --->
BYE sip:3907000@192.168.1.10:5060 SIP/2.0
Via: SIP/2.0/UDP 92.46.61.21:5060;branch=z9hG4bKsgqsib10bo9gnq0c81o1cd1pf1s23.1
From: <sip:87273933837@10.14.0.2;user=phone>;tag=1397113271-1302507520391-
To: "John BABII GALINA VASILEVNA"<sip:133930188999036@sip.telecom.kz;ep=95.56.104.24:5060;fw=95.56.104.24:5060>;tag=as76fce2fb
Call-ID: BW133840391110411-305566094@10.14.0.2
CSeq: 566214853 BYE
Max-Forwards: 69
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Sending to 92.46.61.21 : 5060 (no NAT)

<--- Transmitting (no NAT) to 92.46.61.21:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 92.46.61.21:5060;branch=z9hG4bKsgqsib10bo9gnq0c81o1cd1pf1s23.1;received=92.46.61.21
From: <sip:87273933837@10.14.0.2;user=phone>;tag=1397113271-1302507520391-
To: "John BABII GALINA VASILEVNA"<sip:133930188999036@sip.telecom.kz;ep=95.56.104.24:5060;fw=95.56.104.24:5060>;tag=as76fce2fb
Call-ID: BW133840391110411-305566094@10.14.0.2
CSeq: 566214853 BYE
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
    -- Stopped music on hold on SIP/telecom-3907000-0000000a
    -- Executing [h@macro-dial:1] Macro("SIP/telecom-3907000-0000000a", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/telecom-3907000-0000000a", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,4)
    -- Executing [s@macro-hangupcall:4] GotoIf("SIP/telecom-3907000-0000000a", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/telecom-3907000-0000000a", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] Hangup("SIP/telecom-3907000-0000000a", "") in new stack
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/telecom-3907000-0000000a' in macro 'hangupcall'
  == Spawn extension (macro-dial, s, 7) exited non-zero on 'SIP/telecom-3907000-0000000a' in macro 'dial'
  == Spawn extension (macro-exten-vm, s, 9) exited non-zero on 'SIP/telecom-3907000-0000000a' in macro 'exten-vm'
  == Spawn extension (from-did-direct, 102, 1) exited non-zero on 'SIP/telecom-3907000-0000000a'
Really destroying SIP dialog 'BW133840391110411-305566094@10.14.0.2' Method: BYE
trixbox1*CLI> sip set debug off
SIP Debugging Disabled
trixbox1*CLI> mc
No such command 'mc' (type 'help mc' for other possible commands)
trixbox1*CLI> exit
[trixbox1.localdomain asterisk]# mc

[trixbox1.localdomain asterisk]#
