Log_JorgeCasas

Asterisk 1.8.20.0, Copyright (C) 1999 - 2012 Digium, Inc. and others.
Created by Mark Spencer <[email protected]>
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 pbxhmo (pid = 17641)
pbxhmo*CLI>
Verbosity is at least 3

pbxhmo*CLI>
<--- SIP read from UDP:10.6.58.195:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.6.58.195:5060;branch=z9hG4bK00B7805377874e178f3
From: <sip:[email protected]:5060;otg=HERTMXMOR_1002>;tag=gK0077c850
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 2711 INVITE
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: <sip:[email protected]:5060>
Supported: timer,100rel
Session-Expires: 1800
Min-SE: 90
Content-Length: 259
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 17433 30551 IN IP4 10.6.58.195
s=SIP Media Capabilities
c=IN IP4 10.6.58.197
t=0 0
m=audio 8434 RTP/AVP 0 18 100
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=sendrecv
a=maxptime:20
<------------->

pbxhmo*CLI>
--- (16 headers 12 lines) ---

pbxhmo*CLI>
Sending to 10.6.58.195:5060 (NAT)
Using INVITE request as basis request - [email protected]

pbxhmo*CLI>
Found peer 'alestra' for '6699156000' from 10.6.58.195:5060

pbxhmo*CLI>
  == Using SIP RTP TOS bits 184

pbxhmo*CLI>
  == Using SIP RTP CoS mark 5

pbxhmo*CLI>
Found RTP audio format 0

pbxhmo*CLI>
Found RTP audio format 18

pbxhmo*CLI>
Found RTP audio format 100
Found audio description format PCMU for ID 0

pbxhmo*CLI>
Found audio description format G729 for ID 18

pbxhmo*CLI>
Found audio description format telephone-event for ID 100

pbxhmo*CLI>
Capabilities: us - 0x100 (g729), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)

pbxhmo*CLI>
Peer audio RTP is at port 10.6.58.197:8434

pbxhmo*CLI>
Looking for 2156 in from-trunk-sip-alestra (domain 10.11.3.9)

pbxhmo*CLI>
list_route: hop: <sip:[email protected]:5060>

<--- Transmitting (NAT) to 10.6.58.195:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.6.58.195:5060;branch=z9hG4bK00B7805377874e178f3;received=10.6.58.195;rport=5060
From: <sip:[email protected]:5060;otg=HERTMXMOR_1002>;tag=gK0077c850
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 2711 INVITE
Server: FPBX-2.8.1(1.8.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Length: 0


<------------>
    -- Executing [2156@from-trunk-sip-alestra:1] Set("SIP/alestra-0000008a", "GROUP()=OUT_2") in new stack
    -- Executing [2156@from-trunk-sip-alestra:2] Goto("SIP/alestra-0000008a", "from-trunk,2156,1") in new stack
    -- Goto (from-trunk,2156,1)
    -- Executing [2156@from-trunk:1] Set("SIP/alestra-0000008a", "__FROM_DID=2156") in new stack
    -- Executing [2156@from-trunk:2] Gosub("SIP/alestra-0000008a", "app-blacklist-check,s,1") in new stack
    -- Executing [s@app-blacklist-check:1] GotoIf("SIP/alestra-0000008a", "0?blacklisted") in new stack
    -- Executing [s@app-blacklist-check:2] Set("SIP/alestra-0000008a", "CALLED_BLACKLIST=1") in new stack
    -- Executing [s@app-blacklist-check:3] Return("SIP/alestra-0000008a", "") in new stack
    -- Executing [2156@from-trunk:3] ExecIf("SIP/alestra-0000008a", "1 ?Set(CALLERID(name)=6699156000)") in new stack
    -- Executing [2156@from-trunk:4] Set("SIP/alestra-0000008a", "__CALLINGPRES_SV=allowed_not_screened") in new stack
    -- Executing [2156@from-trunk:5] Set("SIP/alestra-0000008a", "CALLERPRES()=allowed_not_screened") in new stack
    -- Executing [2156@from-trunk:6] Goto("SIP/alestra-0000008a", "from-did-direct,104,1") in new stack
    -- Goto (from-did-direct,104,1)
    -- Executing [104@from-did-direct:1] Macro("SIP/alestra-0000008a", "exten-vm,novm,104") in new stack
    -- Executing [s@macro-exten-vm:1] Macro("SIP/alestra-0000008a", "user-callerid,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/alestra-0000008a", "AMPUSER=6699156000") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/alestra-0000008a", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/alestra-0000008a", "1?Set(REALCALLERIDNUM=6699156000)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/alestra-0000008a", "AMPUSER=") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/alestra-0000008a", "AMPUSERCIDNAME=") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/alestra-0000008a", "1?report") in new stack
    -- Goto (macro-user-callerid,s,10)
    -- Executing [s@macro-user-callerid:10] GotoIf("SIP/alestra-0000008a", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:11] Set("SIP/alestra-0000008a", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:12] GotoIf("SIP/alestra-0000008a", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [s@macro-user-callerid:19] Set("SIP/alestra-0000008a", "CALLERID(number)=6699156000") in new stack
    -- Executing [s@macro-user-callerid:20] Set("SIP/alestra-0000008a", "CALLERID(name)=6699156000") in new stack
    -- Executing [s@macro-user-callerid:21] NoOp("SIP/alestra-0000008a", "Using CallerID "6699156000" <6699156000>") in new stack
    -- Executing [s@macro-exten-vm:2] Set("SIP/alestra-0000008a", "RingGroupMethod=none") in new stack
    -- Executing [s@macro-exten-vm:3] Set("SIP/alestra-0000008a", "VMBOX=novm") in new stack
    -- Executing [s@macro-exten-vm:4] Set("SIP/alestra-0000008a", "__EXTTOCALL=104") in new stack
    -- Executing [s@macro-exten-vm:5] Set("SIP/alestra-0000008a", "CFUEXT=") in new stack
    -- Executing [s@macro-exten-vm:6] Set("SIP/alestra-0000008a", "CFBEXT=") in new stack
    -- Executing [s@macro-exten-vm:7] Set("SIP/alestra-0000008a", "RT=""") in new stack
    -- Executing [s@macro-exten-vm:8] Macro("SIP/alestra-0000008a", "record-enable,104,IN") in new stack

pbxhmo*CLI>
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/alestra-0000008a", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] ExecIf("SIP/alestra-0000008a", "0?MacroExit()") in new stack

pbxhmo*CLI>
    -- Executing [s@macro-record-enable:5] GotoIf("SIP/alestra-0000008a", "0?Group:OUT") in new stack

pbxhmo*CLI>
    -- Goto (macro-record-enable,s,15)

pbxhmo*CLI>
    -- Executing [s@macro-record-enable:15] GotoIf("SIP/alestra-0000008a", "1?IN") in new stack

pbxhmo*CLI>
    -- Goto (macro-record-enable,s,20)

pbxhmo*CLI>
    -- Executing [s@macro-record-enable:20] ExecIf("SIP/alestra-0000008a", "1?MacroExit()") in new stack

pbxhmo*CLI>
    -- Executing [s@macro-exten-vm:9] Macro("SIP/alestra-0000008a", "dial-one,"",tr,104") in new stack

pbxhmo*CLI>
    -- Executing [s@macro-dial-one:1] Set("SIP/alestra-0000008a", "DEXTEN=104") in new stack

pbxhmo*CLI>
    -- Executing [s@macro-dial-one:2] Set("SIP/alestra-0000008a", "DIALSTATUS_CW=") in new stack

pbxhmo*CLI>
    -- Executing [s@macro-dial-one:3] GosubIf("SIP/alestra-0000008a", "0?screen,1") in new stack

pbxhmo*CLI>
    -- Executing [s@macro-dial-one:4] GosubIf("SIP/alestra-0000008a", "0?cf,1") in new stack

pbxhmo*CLI>
    -- Executing [s@macro-dial-one:5] GotoIf("SIP/alestra-0000008a", "1?skip1") in new stack

pbxhmo*CLI>
    -- Goto (macro-dial-one,s,8)

pbxhmo*CLI>
    -- Executing [s@macro-dial-one:8] GotoIf("SIP/alestra-0000008a", "0?nodial") in new stack

pbxhmo*CLI>
    -- Executing [s@macro-dial-one:9] GotoIf("SIP/alestra-0000008a", "0?continue") in new stack

pbxhmo*CLI>
    -- Executing [s@macro-dial-one:10] Set("SIP/alestra-0000008a", "EXTHASCW=") in new stack

pbxhmo*CLI>
    -- Executing [s@macro-dial-one:11] GotoIf("SIP/alestra-0000008a", "1?next1:cwinusebusy") in new stack

pbxhmo*CLI>
    -- Goto (macro-dial-one,s,12)

pbxhmo*CLI>
    -- Executing [s@macro-dial-one:12] GotoIf("SIP/alestra-0000008a", "0?docfu:skip3") in new stack

pbxhmo*CLI>
    -- Goto (macro-dial-one,s,16)

pbxhmo*CLI>
    -- Executing [s@macro-dial-one:16] GotoIf("SIP/alestra-0000008a", "1?next2:continue") in new stack

pbxhmo*CLI>
    -- Goto (macro-dial-one,s,17)

pbxhmo*CLI>
    -- Executing [s@macro-dial-one:17] GotoIf("SIP/alestra-0000008a", "1?continue") in new stack

pbxhmo*CLI>
    -- Goto (macro-dial-one,s,25)

pbxhmo*CLI>
    -- Executing [s@macro-dial-one:25] GotoIf("SIP/alestra-0000008a", "0?nodial") in new stack

pbxhmo*CLI>
    -- Executing [s@macro-dial-one:26] GosubIf("SIP/alestra-0000008a", "1?dstring,1:dlocal,1") in new stack

pbxhmo*CLI>
    -- Executing [dstring@macro-dial-one:1] Set("SIP/alestra-0000008a", "DSTRING=") in new stack

pbxhmo*CLI>
    -- Executing [dstring@macro-dial-one:2] Set("SIP/alestra-0000008a", "DEVICES=104") in new stack

pbxhmo*CLI>
    -- Executing [dstring@macro-dial-one:3] ExecIf("SIP/alestra-0000008a", "0?Return()") in new stack

pbxhmo*CLI>
    -- Executing [dstring@macro-dial-one:4] ExecIf("SIP/alestra-0000008a", "0?Set(DEVICES=04)") in new stack

pbxhmo*CLI>
    -- Executing [dstring@macro-dial-one:5] Set("SIP/alestra-0000008a", "LOOPCNT=1") in new stack

pbxhmo*CLI>
    -- Executing [dstring@macro-dial-one:6] Set("SIP/alestra-0000008a", "ITER=1") in new stack

pbxhmo*CLI>
    -- Executing [dstring@macro-dial-one:7] Set("SIP/alestra-0000008a", "THISDIAL=SIP/104") in new stack

pbxhmo*CLI>
    -- Executing [dstring@macro-dial-one:8] GosubIf("SIP/alestra-0000008a", "1?zap2dahdi,1") in new stack

pbxhmo*CLI>
    -- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/alestra-0000008a", "0?Return()") in new stack

pbxhmo*CLI>
    -- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/alestra-0000008a", "NEWDIAL=") in new stack

pbxhmo*CLI>
    -- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/alestra-0000008a", "LOOPCNT2=1") in new stack

pbxhmo*CLI>
    -- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/alestra-0000008a", "ITER2=1") in new stack

pbxhmo*CLI>
    -- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/alestra-0000008a", "THISPART2=SIP/104") in new stack

pbxhmo*CLI>
    -- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/alestra-0000008a", "0?Set(THISPART2=DAHDI/104)") in new stack

pbxhmo*CLI>
    -- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/alestra-0000008a", "NEWDIAL=SIP/104&") in new stack

pbxhmo*CLI>
    -- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/alestra-0000008a", "ITER2=2") in new stack

pbxhmo*CLI>
    -- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/alestra-0000008a", "0?begin2") in new stack

pbxhmo*CLI>
    -- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/alestra-0000008a", "THISDIAL=SIP/104") in new stack

pbxhmo*CLI>
    -- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/alestra-0000008a", "") in new stack

pbxhmo*CLI>
    -- Executing [dstring@macro-dial-one:9] Set("SIP/alestra-0000008a", "DSTRING=SIP/104&") in new stack

pbxhmo*CLI>
    -- Executing [dstring@macro-dial-one:10] Set("SIP/alestra-0000008a", "ITER=2") in new stack

pbxhmo*CLI>
    -- Executing [dstring@macro-dial-one:11] GotoIf("SIP/alestra-0000008a", "0?begin") in new stack

pbxhmo*CLI>
    -- Executing [dstring@macro-dial-one:12] Set("SIP/alestra-0000008a", "DSTRING=SIP/104") in new stack

pbxhmo*CLI>
    -- Executing [dstring@macro-dial-one:13] Return("SIP/alestra-0000008a", "") in new stack

pbxhmo*CLI>
    -- Executing [s@macro-dial-one:27] GotoIf("SIP/alestra-0000008a", "0?nodial") in new stack

pbxhmo*CLI>
    -- Executing [s@macro-dial-one:28] GotoIf("SIP/alestra-0000008a", "0?skiptrace") in new stack

pbxhmo*CLI>
    -- Executing [s@macro-dial-one:29] GosubIf("SIP/alestra-0000008a", "1?ctset,1:ctclear,1") in new stack

pbxhmo*CLI>
    -- Executing [ctset@macro-dial-one:1] Set("SIP/alestra-0000008a", "DB(CALLTRACE/104)=6699156000") in new stack

pbxhmo*CLI>
    -- Executing [ctset@macro-dial-one:2] Return("SIP/alestra-0000008a", "") in new stack

pbxhmo*CLI>
    -- Executing [s@macro-dial-one:30] Set("SIP/alestra-0000008a", "D_OPTIONS=tr") in new stack

pbxhmo*CLI>
    -- Executing [s@macro-dial-one:31] ExecIf("SIP/alestra-0000008a", "0?SIPAddHeader(Alert-Info: )") in new stack

pbxhmo*CLI>
    -- Executing [s@macro-dial-one:32] ExecIf("SIP/alestra-0000008a", "0?SIPAddHeader()") in new stack

pbxhmo*CLI>
    -- Executing [s@macro-dial-one:33] ExecIf("SIP/alestra-0000008a", "0?Set(CHANNEL(musicclass)=)") in new stack

pbxhmo*CLI>
    -- Executing [s@macro-dial-one:34] GosubIf("SIP/alestra-0000008a", "0?qwait,1") in new stack

pbxhmo*CLI>
    -- Executing [s@macro-dial-one:35] Set("SIP/alestra-0000008a", "__CWIGNORE=") in new stack

pbxhmo*CLI>
    -- Executing [s@macro-dial-one:36] Set("SIP/alestra-0000008a", "__KEEPCID=TRUE") in new stack

pbxhmo*CLI>
    -- Executing [s@macro-dial-one:37] Dial("SIP/alestra-0000008a", "SIP/104,"",tr") in new stack

pbxhmo*CLI>
  == Using SIP RTP TOS bits 184

pbxhmo*CLI>
  == Using SIP RTP CoS mark 5

pbxhmo*CLI>
Audio is at 13306

pbxhmo*CLI>
Adding codec 0x4 (ulaw) to SDP

pbxhmo*CLI>
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP

pbxhmo*CLI>
Adding non-codec 0x1 (telephone-event) to SDP

pbxhmo*CLI>
Reliably Transmitting (NAT) to 10.11.1.153:5062:
INVITE sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/UDP 10.11.1.9:5060;branch=z9hG4bK30946f68;rport
Max-Forwards: 70
From: "6699156000" <sip:[email protected]>;tag=as35a7e686
To: <sip:[email protected]:5062>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.8.20.0)
Date: Tue, 31 Dec 2013 20:17:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 276

v=0
o=root 241784902 241784902 IN IP4 10.11.1.9
s=Asterisk PBX 1.8.20.0
c=IN IP4 10.11.1.9
t=0 0
m=audio 13306 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

pbxhmo*CLI>
    -- Called SIP/104

pbxhmo*CLI>
<--- Transmitting (NAT) to 10.6.58.195:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.6.58.195:5060;branch=z9hG4bK00B7805377874e178f3;received=10.6.58.195;rport=5060
From: <sip:[email protected]:5060;otg=HERTMXMOR_1002>;tag=gK0077c850
To: <sip:[email protected]:5060>;tag=as686231eb
Call-ID: [email protected]
CSeq: 2711 INVITE
Server: FPBX-2.8.1(1.8.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Length: 0


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

pbxhmo*CLI>
<--- SIP read from UDP:10.11.1.153:5062 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.11.1.9:5060;branch=z9hG4bK30946f68;rport
From: "6699156000" <sip:[email protected]>;tag=as35a7e686
To: <sip:[email protected]:5062>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Yealink SIP-T20P 9.71.0.168
Content-Length: 0

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

pbxhmo*CLI>
--- (8 headers 0 lines) ---

pbxhmo*CLI>
<--- SIP read from UDP:10.11.1.153:5062 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.11.1.9:5060;branch=z9hG4bK30946f68;rport
From: "6699156000" <sip:[email protected]>;tag=as35a7e686
To: <sip:[email protected]:5062>;tag=973490739
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Contact: <sip:[email protected]:5062>
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
User-Agent: Yealink SIP-T20P 9.71.0.168
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 0

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

pbxhmo*CLI>
--- (11 headers 0 lines) ---

pbxhmo*CLI>
list_route: hop: <sip:[email protected]:5062>

pbxhmo*CLI>
    -- SIP/104-0000008b is ringing

pbxhmo*CLI>
<--- Transmitting (NAT) to 10.6.58.195:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.6.58.195:5060;branch=z9hG4bK00B7805377874e178f3;received=10.6.58.195;rport=5060
From: <sip:[email protected]:5060;otg=HERTMXMOR_1002>;tag=gK0077c850
To: <sip:[email protected]:5060>;tag=as686231eb
Call-ID: [email protected]
CSeq: 2711 INVITE
Server: FPBX-2.8.1(1.8.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Length: 0


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

pbxhmo*CLI>
Reliably Transmitting (NAT) to 10.11.1.156:5062:
OPTIONS sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/UDP 10.11.1.9:5060;branch=z9hG4bK4de87984;rport
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=as68bcda31
To: <sip:[email protected]:5062>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.8.1(1.8.20.0)
Date: Tue, 31 Dec 2013 20:17:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

pbxhmo*CLI>
<--- SIP read from UDP:10.11.1.156:5062 --->


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

pbxhmo*CLI>
<--- SIP read from UDP:10.11.1.156:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.11.1.9:5060;branch=z9hG4bK4de87984;rport
From: "Unknown" <sip:[email protected]>;tag=as68bcda31
To: <sip:[email protected]:5062>;tag=1463935576
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Yealink SIP-T20P 9.71.0.168
Content-Length: 0

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

pbxhmo*CLI>
--- (8 headers 0 lines) ---

pbxhmo*CLI>
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

pbxhmo*CLI>
Reliably Transmitting (NAT) to 10.11.1.159:5062:
OPTIONS sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/UDP 10.11.1.9:5060;branch=z9hG4bK552f4af7;rport
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=as5b4d05c7
To: <sip:[email protected]:5062>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.8.1(1.8.20.0)
Date: Tue, 31 Dec 2013 20:17:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

pbxhmo*CLI>
<--- SIP read from UDP:10.11.1.159:5062 --->


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

pbxhmo*CLI>
<--- SIP read from UDP:10.11.1.159:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.11.1.9:5060;branch=z9hG4bK552f4af7;rport
From: "Unknown" <sip:[email protected]>;tag=as5b4d05c7
To: <sip:[email protected]:5062>;tag=403368631
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Yealink SIP-T20P 9.71.0.168
Content-Length: 0

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

pbxhmo*CLI>
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

pbxhmo*CLI>
Reliably Transmitting (NAT) to 10.11.1.164:5062:
OPTIONS sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/UDP 10.11.1.9:5060;branch=z9hG4bK180b13e9;rport
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=as7e7435fe
To: <sip:[email protected]:5062>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.8.1(1.8.20.0)
Date: Tue, 31 Dec 2013 20:17:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

pbxhmo*CLI>
<--- SIP read from UDP:10.11.1.164:5062 --->


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

pbxhmo*CLI>
<--- SIP read from UDP:10.11.1.164:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.11.1.9:5060;branch=z9hG4bK180b13e9;rport
From: "Unknown" <sip:[email protected]>;tag=as7e7435fe
To: <sip:[email protected]:5062>;tag=1752043124
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Yealink SIP-T20P 9.71.0.168
Content-Length: 0

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

pbxhmo*CLI>
--- (8 headers 0 lines) ---

pbxhmo*CLI>
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

pbxhmo*CLI>
<--- SIP read from UDP:10.11.1.154:5062 --->


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

pbxhmo*CLI>
<--- SIP read from UDP:10.6.58.195:5060 --->
CANCEL sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.6.58.195:5060;branch=z9hG4bK00B7805377874e178f3
From: <sip:[email protected]:5060;otg=HERTMXMOR_1002>;tag=gK0077c850
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 2711 CANCEL
Max-Forwards: 70
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Sending to 10.6.58.195:5060 (NAT)

<--- Reliably Transmitting (NAT) to 10.6.58.195:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.6.58.195:5060;branch=z9hG4bK00B7805377874e178f3;received=10.6.58.195;rport=5060
From: <sip:[email protected]:5060;otg=HERTMXMOR_1002>;tag=gK0077c850
To: <sip:[email protected]:5060>;tag=as686231eb
Call-ID: [email protected]
CSeq: 2711 INVITE
Server: FPBX-2.8.1(1.8.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


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

pbxhmo*CLI>
<--- Transmitting (NAT) to 10.6.58.195:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.6.58.195:5060;branch=z9hG4bK00B7805377874e178f3;received=10.6.58.195;rport=5060
From: <sip:[email protected]:5060;otg=HERTMXMOR_1002>;tag=gK0077c850
To: <sip:[email protected]:5060>;tag=as686231eb
Call-ID: [email protected]
CSeq: 2711 CANCEL
Server: FPBX-2.8.1(1.8.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]:5060' in 6400 ms (Method: INVITE)
Reliably Transmitting (NAT) to 10.11.1.153:5062:
CANCEL sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/UDP 10.11.1.9:5060;branch=z9hG4bK30946f68;rport
Max-Forwards: 70
From: "6699156000" <sip:[email protected]>;tag=as35a7e686
To: <sip:[email protected]:5062>
Call-ID: [email protected]:5060
CSeq: 102 CANCEL
User-Agent: FPBX-2.8.1(1.8.20.0)
Content-Length: 0


---
Scheduling destruction of SIP dialog '[email protected]:5060' in 6400 ms (Method: INVITE)

pbxhmo*CLI>
  == Spawn extension (macro-dial-one, s, 37) exited non-zero on 'SIP/alestra-0000008a' in macro 'dial-one'
  == Spawn extension (macro-exten-vm, s, 9) exited non-zero on 'SIP/alestra-0000008a' in macro 'exten-vm'
  == Spawn extension (from-did-direct, 104, 1) exited non-zero on 'SIP/alestra-0000008a'
    -- Executing [h@from-did-direct:1] Macro("SIP/alestra-0000008a", "hangupcall,") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/alestra-0000008a", "1?endmixmoncheck") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] NoOp("SIP/alestra-0000008a", "End of MIXMON check") in new stack
    -- Executing [s@macro-hangupcall:10] GotoIf("SIP/alestra-0000008a", "1?nomeetmemon") in new stack

pbxhmo*CLI>
    -- Goto (macro-hangupcall,s,28)

pbxhmo*CLI>
    -- Executing [s@macro-hangupcall:28] NoOp("SIP/alestra-0000008a", "End of MEETME check") in new stack

pbxhmo*CLI>
    -- Executing [s@macro-hangupcall:29] GotoIf("SIP/alestra-0000008a", "1?noautomon") in new stack

pbxhmo*CLI>
    -- Goto (macro-hangupcall,s,34)

pbxhmo*CLI>
    -- Executing [s@macro-hangupcall:34] NoOp("SIP/alestra-0000008a", "TOUCH_MONITOR_OUTPUT=") in new stack

pbxhmo*CLI>
    -- Executing [s@macro-hangupcall:35] GotoIf("SIP/alestra-0000008a", "1?noautomon2") in new stack

pbxhmo*CLI>
    -- Goto (macro-hangupcall,s,41)

pbxhmo*CLI>
    -- Executing [s@macro-hangupcall:41] NoOp("SIP/alestra-0000008a", "MONITOR_FILENAME=") in new stack

pbxhmo*CLI>
    -- Executing [s@macro-hangupcall:42] GotoIf("SIP/alestra-0000008a", "1?skiprg") in new stack

pbxhmo*CLI>
    -- Goto (macro-hangupcall,s,45)

pbxhmo*CLI>
    -- Executing [s@macro-hangupcall:45] GotoIf("SIP/alestra-0000008a", "1?skipblkvm") in new stack

pbxhmo*CLI>
    -- Goto (macro-hangupcall,s,48)

pbxhmo*CLI>
    -- Executing [s@macro-hangupcall:48] GotoIf("SIP/alestra-0000008a", "1?theend") in new stack

pbxhmo*CLI>
    -- Goto (macro-hangupcall,s,50)

pbxhmo*CLI>
    -- Executing [s@macro-hangupcall:50] AGI("SIP/alestra-0000008a", "hangup.agi") in new stack

pbxhmo*CLI>
    -- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi

pbxhmo*CLI>
<--- SIP read from UDP:10.11.1.153:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.11.1.9:5060;branch=z9hG4bK30946f68;rport
From: "6699156000" <sip:[email protected]>;tag=as35a7e686
To: <sip:[email protected]:5062>;tag=973490739
Call-ID: [email protected]:5060
CSeq: 102 CANCEL
User-Agent: Yealink SIP-T20P 9.71.0.168
Content-Length: 0

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

pbxhmo*CLI>
<--- SIP read from UDP:10.11.1.153:5062 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.11.1.9:5060;branch=z9hG4bK30946f68;rport
From: "6699156000" <sip:[email protected]>;tag=as35a7e686
To: <sip:[email protected]:5062>;tag=973490739
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Yealink SIP-T20P 9.71.0.168
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 10.11.1.153:5062:
ACK sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/UDP 10.11.1.9:5060;branch=z9hG4bK30946f68;rport
Max-Forwards: 70
From: "6699156000" <sip:[email protected]>;tag=as35a7e686
To: <sip:[email protected]:5062>;tag=973490739
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: FPBX-2.8.1(1.8.20.0)
Content-Length: 0


---
Scheduling destruction of SIP dialog '[email protected]:5060' in 6400 ms (Method: INVITE)

pbxhmo*CLI>
<--- SIP read from UDP:10.6.58.195:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.6.58.195:5060;branch=z9hG4bK00B7805377874e178f3
From: <sip:[email protected]:5060;otg=HERTMXMOR_1002>;tag=gK0077c850
To: <sip:[email protected]:5060>;tag=as686231eb
Call-ID: [email protected]
CSeq: 2711 ACK
Max-Forwards: 70
Content-Length: 0

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

pbxhmo*CLI>
--- (8 headers 0 lines) ---

pbxhmo*CLI>
    -- <SIP/alestra-0000008a>AGI Script hangup.agi completed, returning 0

pbxhmo*CLI>
    -- Executing [s@macro-hangupcall:51] Hangup("SIP/alestra-0000008a", "") in new stack

pbxhmo*CLI>
  == Spawn extension (macro-hangupcall, s, 51) exited non-zero on 'SIP/alestra-0000008a' in macro 'hangupcall'

pbxhmo*CLI>
  == Spawn extension (from-did-direct, h, 1) exited non-zero on 'SIP/alestra-0000008a'

pbxhmo*CLI>
Really destroying SIP dialog '[email protected]' Method: ACK

pbxhmo*CLI>
<--- SIP read from UDP:10.11.1.158:5062 --->


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

pbxhmo*CLI>
Disconnected from Asterisk server
Executing last minute cleanups