Last login: Tue Sep 29 13:02:49 2009 from 192.168.254.3

Welcome to trixbox CE
-------------------------------------------------

For access to the trixbox web GUI use this URL

 br0 http://192.168.254.254
ppp0 http://95.65.180.146


For help on trixbox commands you can use from this
command shell type help-trixbox.

[trixbox1.localdomain ~]# !ast
asterisk -vvvvvvvvvr
Asterisk 1.6.0.10-FONCORE-r40, Copyright (C) 1999 - 2008 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.
=========================================================================
  == Parsing '/etc/asterisk/asterisk.conf':   == Found
  == Parsing '/etc/asterisk/extconfig.conf':   == Found
Connected to Asterisk 1.6.0.10-FONCORE-r40 currently running on trixbox1 (pid = 2930)
Verbosity is at least 9
trixbox1*CLI> sip show history 
Usage: sip show history <call-id>
       Provides detailed dialog history on a given SIP call (specified by call-id).
trixbox1*CLI> sip set debug 
No such command 'sip set debug ' (type 'help sip set debug' for other possible commands)
trixbox1*CLI> sip set debug on
SIP Debugging enabled
Reliably Transmitting (no NAT) to 192.168.254.11:5060:
OPTIONS sip:999@192.168.254.11:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.254.254:5060;branch=z9hG4bK343fc7fc;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.254.254>;tag=as42884db3
To: <sip:999@192.168.254.11:5060>
Contact: <sip:Unknown@192.168.254.254>
Call-ID: 38af807778863f4f6d4bc5ed5be2b86e@192.168.254.254
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Date: Tue, 29 Sep 2009 16:09:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0


---
trixbox1*CLI> 
<--- SIP read from UDP://192.168.254.11:5060 --->
SIP/2.0 486 Busy Here
To: <sip:999@192.168.254.11:5060>;tag=31df190ad9bcde24i0
From: "Unknown" <sip:Unknown@192.168.254.254>;tag=as42884db3
Call-ID: 38af807778863f4f6d4bc5ed5be2b86e@192.168.254.254
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.254.254:5060;branch=z9hG4bK343fc7fc
Server: Linksys/PAP2-3.1.22(LS)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces


<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '38af807778863f4f6d4bc5ed5be2b86e@192.168.254.254' Method: OPTIONS
Reliably Transmitting (no NAT) to 192.168.254.5:5060:
OPTIONS sip:995@192.168.254.5:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.254.254:5060;branch=z9hG4bK0733e499;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.254.254>;tag=as55d50afe
To: <sip:995@192.168.254.5:5060>
Contact: <sip:Unknown@192.168.254.254>
Call-ID: 17816d642fdec511058b7cdf22b6e979@192.168.254.254
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Date: Tue, 29 Sep 2009 16:09:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0


---
trixbox1*CLI> 
<--- SIP read from UDP://192.168.254.5:5060 --->
SIP/2.0 200 OK
To: <sip:995@192.168.254.5:5060>;tag=e8621188cecb1025i0
From: "Unknown" <sip:Unknown@192.168.254.254>;tag=as55d50afe
Call-ID: 17816d642fdec511058b7cdf22b6e979@192.168.254.254
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.254.254:5060;branch=z9hG4bK0733e499;rport=5060
Server: Linksys/SPA3000-3.1.10(GWd)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura


<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '17816d642fdec511058b7cdf22b6e979@192.168.254.254' Method: OPTIONS
trixbox1*CLI> 
<--- SIP read from UDP://192.168.254.11:5060 --->
INVITE sip:995@192.168.254.254 SIP/2.0
Via: SIP/2.0/UDP 192.168.254.11:5060;branch=z9hG4bK-406ae79e
From: ilker <sip:999@192.168.254.254>;tag=a5b4e3ee2377931eo0
To: <sip:995@192.168.254.254>
Call-ID: 41fb902e-f8e6d3de@192.168.254.11
CSeq: 101 INVITE
Max-Forwards: 70
Contact: ilker <sip:999@192.168.254.11:5060>
Expires: 240
User-Agent: Linksys/PAP2-3.1.22(LS)
Content-Length: 452
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp

v=0
o=- 148022197 148022197 IN IP4 192.168.254.11
s=-
c=IN IP4 192.168.254.11
t=0 0
m=audio 16392 RTP/AVP 18 0 2 4 8 96 97 98 100 101
a=rtpmap:18 G729a/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
--- (14 headers 20 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 192.168.254.11 : 5060 (no NAT)
Using INVITE request as basis request - 41fb902e-f8e6d3de@192.168.254.11
Found user '999' for '999'

<--- Reliably Transmitting (no NAT) to 192.168.254.11:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.254.11:5060;branch=z9hG4bK-406ae79e;received=192.168.254.11
From: ilker <sip:999@192.168.254.254>;tag=a5b4e3ee2377931eo0
To: <sip:995@192.168.254.254>;tag=as1323aa9f
Call-ID: 41fb902e-f8e6d3de@192.168.254.11
CSeq: 101 INVITE
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="501b1064"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '41fb902e-f8e6d3de@192.168.254.11' in 32000 ms (Method: INVITE)
trixbox1*CLI> 
<--- SIP read from UDP://192.168.254.11:5060 --->
ACK sip:995@192.168.254.254 SIP/2.0
Via: SIP/2.0/UDP 192.168.254.11:5060;branch=z9hG4bK-406ae79e
From: ilker <sip:999@192.168.254.254>;tag=a5b4e3ee2377931eo0
To: <sip:995@192.168.254.254>;tag=as1323aa9f
Call-ID: 41fb902e-f8e6d3de@192.168.254.11
CSeq: 101 ACK
Max-Forwards: 70
Contact: ilker <sip:999@192.168.254.11:5060>
User-Agent: Linksys/PAP2-3.1.22(LS)
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
trixbox1*CLI> 
<--- SIP read from UDP://192.168.254.11:5060 --->
INVITE sip:995@192.168.254.254 SIP/2.0
Via: SIP/2.0/UDP 192.168.254.11:5060;branch=z9hG4bK-e34764ba
From: ilker <sip:999@192.168.254.254>;tag=a5b4e3ee2377931eo0
To: <sip:995@192.168.254.254>
Call-ID: 41fb902e-f8e6d3de@192.168.254.11
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest username="999",realm="asterisk",nonce="501b1064",uri="sip:995@192.168.254.254",algorithm=MD5,response="09f395f4a4cd35a945427ee18e40ae5e"
Contact: ilker <sip:999@192.168.254.11:5060>
Expires: 240
User-Agent: Linksys/PAP2-3.1.22(LS)
Content-Length: 452
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp

v=0
o=- 148022197 148022197 IN IP4 192.168.254.11
s=-
c=IN IP4 192.168.254.11
t=0 0
m=audio 16392 RTP/AVP 18 0 2 4 8 96 97 98 100 101
a=rtpmap:18 G729a/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
--- (15 headers 20 lines) ---
Sending to 192.168.254.11 : 5060 (no NAT)
Using INVITE request as basis request - 41fb902e-f8e6d3de@192.168.254.11
Found user '999' for '999'
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 192.168.254.11:16392
Found audio description format G729a for ID 18
Found audio description format PCMU for ID 0
Found audio description format G726-32 for ID 2
Found audio description format G723 for ID 4
Found audio description format PCMA for ID 8
Found unknown media description format G726-40 for ID 96
Found unknown media description format G726-24 for ID 97
Found unknown media description format G726-16 for ID 98
Found unknown media description format NSE for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x90d (g723|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.254.11:16392
Looking for 995 in access (domain 192.168.254.254)
list_route: hop: <sip:999@192.168.254.11:5060>
trixbox1*CLI> 
<--- Transmitting (no NAT) to 192.168.254.11:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.254.11:5060;branch=z9hG4bK-e34764ba;received=192.168.254.11
From: ilker <sip:999@192.168.254.254>;tag=a5b4e3ee2377931eo0
To: <sip:995@192.168.254.254>
Call-ID: 41fb902e-f8e6d3de@192.168.254.11
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:995@192.168.254.254>
Content-Length: 0


<------------>
    -- Executing [995@access:1] Macro("SIP/999-b5a03268", "exten-vm,995,995") in new stack
    -- Executing [s@macro-exten-vm:1] Macro("SIP/999-b5a03268", "user-callerid") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/999-b5a03268", "AMPUSER=999") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/999-b5a03268", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/999-b5a03268", "1?Set(REALCALLERIDNUM=999)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/999-b5a03268", "AMPUSER=999") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/999-b5a03268", "AMPUSERCIDNAME=Oda") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/999-b5a03268", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/999-b5a03268", "AMPUSERCID=999") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/999-b5a03268", "CALLERID(all)="Oda" <999>") in new stack
    -- Executing [s@macro-user-callerid:9] Set("SIP/999-b5a03268", "REALCALLERIDNUM=999") in new stack
    -- Executing [s@macro-user-callerid:10] ExecIf("SIP/999-b5a03268", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [s@macro-user-callerid:11] GotoIf("SIP/999-b5a03268", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:12] Set("SIP/999-b5a03268", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:13] GotoIf("SIP/999-b5a03268", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,20)
    -- Executing [s@macro-user-callerid:20] NoOp("SIP/999-b5a03268", "Using CallerID "Oda" <999>") in new stack
    -- Executing [s@macro-exten-vm:2] Set("SIP/999-b5a03268", "RingGroupMethod=none") in new stack
    -- Executing [s@macro-exten-vm:3] Set("SIP/999-b5a03268", "VMBOX=995") in new stack
    -- Executing [s@macro-exten-vm:4] Set("SIP/999-b5a03268", "EXTTOCALL=995") in new stack
    -- Executing [s@macro-exten-vm:5] Set("SIP/999-b5a03268", "CFUEXT=") in new stack
    -- Executing [s@macro-exten-vm:6] Set("SIP/999-b5a03268", "CFBEXT=") in new stack
    -- Executing [s@macro-exten-vm:7] Set("SIP/999-b5a03268", "RT=15") in new stack
    -- Executing [s@macro-exten-vm:8] Macro("SIP/999-b5a03268", "record-enable,995,IN") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/999-b5a03268", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] AGI("SIP/999-b5a03268", "recordingcheck,20090929-190956,1254240596.158") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
 recordingcheck,20090929-190956,1254240596.158: Inbound recording enabled.
 recordingcheck,20090929-190956,1254240596.158: CALLFILENAME=20090929-190956-1254240596.158
    -- <SIP/999-b5a03268>AGI Script recordingcheck completed, returning 0
    -- Executing [s@macro-record-enable:999] MixMonitor("SIP/999-b5a03268", "20090929-190956-1254240596.158.wav,,") in new stack
  == Begin MixMonitor Recording SIP/999-b5a03268
    -- Executing [s@macro-exten-vm:9] Macro("SIP/999-b5a03268", "dial,15,tr,995") in new stack
    -- Executing [s@macro-dial:1] GotoIf("SIP/999-b5a03268", "1?dial") in new stack
    -- Goto (macro-dial,s,3)
    -- Executing [s@macro-dial:3] AGI("SIP/999-b5a03268", "dialparties.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
 dialparties.agi: Starting New Dialparties.agi
  == Manager 'admin' logged on from 127.0.0.1
 dialparties.agi: Caller ID name is 'Oda' number is '999'
       > dialparties.agi: USE_CONFIRMATION:  'FALSE'
       > dialparties.agi: RINGGROUP_INDEX:   ''
 dialparties.agi: Methodology of ring is  'none'
    -- dialparties.agi: Added extension 995 to extension map
    -- dialparties.agi: Extension 995 cf is disabled
    -- dialparties.agi: Extension 995 do not disturb is disabled
       > dialparties.agi: extnum 995 has:  cw: 1; hascfb: 0 [] hascfu: 0 []
       > dialparties.agi: ExtensionState: 0
    -- dialparties.agi: dbset CALLTRACE/995 to 999
    -- dialparties.agi: Filtered ARG3: 995
  == Manager 'admin' logged off from 127.0.0.1
    -- <SIP/999-b5a03268>AGI Script dialparties.agi completed, returning 0
    -- Executing [s@macro-dial:7] Dial("SIP/999-b5a03268", "SIP/995,15,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
Audio is at 192.168.254.254 port 32930
Adding codec 0x8 (alaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x4 (ulaw) to SDP
Reliably Transmitting (no NAT) to 192.168.254.5:5060:
INVITE sip:995@192.168.254.5:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.254.254:5060;branch=z9hG4bK36ff9c02;rport
Max-Forwards: 70
From: "Oda" <sip:999@192.168.254.254>;tag=as0c63ad03
To: <sip:995@192.168.254.5:5060>
Contact: <sip:999@192.168.254.254>
Call-ID: 4ecacd4a676f918d1f46b833522ee12d@192.168.254.254
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Date: Tue, 29 Sep 2009 16:10:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 297

v=0
o=root 1010444261 1010444261 IN IP4 192.168.254.254
s=Asterisk PBX 1.6.0.10-FONCORE-r40
c=IN IP4 192.168.254.254
t=0 0
m=audio 32930 RTP/AVP 8 18 0
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 995
trixbox1*CLI> 
<--- SIP read from UDP://192.168.254.5:5060 --->
SIP/2.0 100 Trying
To: <sip:995@192.168.254.5:5060>
From: "Oda" <sip:999@192.168.254.254>;tag=as0c63ad03
Call-ID: 4ecacd4a676f918d1f46b833522ee12d@192.168.254.254
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.254.254:5060;branch=z9hG4bK36ff9c02;rport=5060
Server: Linksys/SPA3000-3.1.10(GWd)
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
trixbox1*CLI> 
<--- Transmitting (no NAT) to 192.168.254.11:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.254.11:5060;branch=z9hG4bK-e34764ba;received=192.168.254.11
From: ilker <sip:999@192.168.254.254>;tag=a5b4e3ee2377931eo0
To: <sip:995@192.168.254.254>;tag=as1ba4eab0
Call-ID: 41fb902e-f8e6d3de@192.168.254.11
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:995@192.168.254.254>
Content-Length: 0


<------------>
trixbox1*CLI> 
<--- SIP read from UDP://192.168.254.5:5060 --->
SIP/2.0 180 Ringing
To: <sip:995@192.168.254.5:5060>;tag=351248c225334577i0
From: "Oda" <sip:999@192.168.254.254>;tag=as0c63ad03
Call-ID: 4ecacd4a676f918d1f46b833522ee12d@192.168.254.254
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.254.254:5060;branch=z9hG4bK36ff9c02;rport=5060
Server: Linksys/SPA3000-3.1.10(GWd)
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
    -- SIP/995-084e5d08 is ringing
trixbox1*CLI> 
<--- Transmitting (no NAT) to 192.168.254.11:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.254.11:5060;branch=z9hG4bK-e34764ba;received=192.168.254.11
From: ilker <sip:999@192.168.254.254>;tag=a5b4e3ee2377931eo0
To: <sip:995@192.168.254.254>;tag=as1ba4eab0
Call-ID: 41fb902e-f8e6d3de@192.168.254.11
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:995@192.168.254.254>
Content-Length: 0


<------------>
    -- Nobody picked up in 15000 ms
Scheduling destruction of SIP dialog '4ecacd4a676f918d1f46b833522ee12d@192.168.254.254' in 6400 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 192.168.254.5:5060:
CANCEL sip:995@192.168.254.5:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.254.254:5060;branch=z9hG4bK36ff9c02;rport
Max-Forwards: 70
From: "Oda" <sip:999@192.168.254.254>;tag=as0c63ad03
To: <sip:995@192.168.254.5:5060>
Call-ID: 4ecacd4a676f918d1f46b833522ee12d@192.168.254.254
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Content-Length: 0


---
Scheduling destruction of SIP dialog '4ecacd4a676f918d1f46b833522ee12d@192.168.254.254' in 6400 ms (Method: INVITE)
    -- Executing [s@macro-dial:8] Set("SIP/999-b5a03268", "DIALSTATUS=NOANSWER") in new stack
    -- Executing [s@macro-dial:9] GosubIf("SIP/999-b5a03268", "0?NOANSWER,1") in new stack
trixbox1*CLI> 
<--- SIP read from UDP://192.168.254.5:5060 --->
SIP/2.0 487 Request Terminated
To: <sip:995@192.168.254.5:5060>;tag=351248c225334577i0
From: "Oda" <sip:999@192.168.254.254>;tag=as0c63ad03
Call-ID: 4ecacd4a676f918d1f46b833522ee12d@192.168.254.254
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.254.254:5060;branch=z9hG4bK36ff9c02;rport=5060
Server: Linksys/SPA3000-3.1.10(GWd)
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 192.168.254.5:5060:
ACK sip:995@192.168.254.5:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.254.254:5060;branch=z9hG4bK36ff9c02;rport
Max-Forwards: 70
From: "Oda" <sip:999@192.168.254.254>;tag=as0c63ad03
To: <sip:995@192.168.254.5:5060>;tag=351248c225334577i0
Contact: <sip:999@192.168.254.254>
Call-ID: 4ecacd4a676f918d1f46b833522ee12d@192.168.254.254
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Content-Length: 0


---

<--- SIP read from UDP://192.168.254.5:5060 --->
SIP/2.0 200 OK
To: <sip:995@192.168.254.5:5060>;tag=351248c225334577i0
From: "Oda" <sip:999@192.168.254.254>;tag=as0c63ad03
Call-ID: 4ecacd4a676f918d1f46b833522ee12d@192.168.254.254
CSeq: 102 CANCEL
Via: SIP/2.0/UDP 192.168.254.254:5060;branch=z9hG4bK36ff9c02;rport=5060
Server: Linksys/SPA3000-3.1.10(GWd)
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '4ecacd4a676f918d1f46b833522ee12d@192.168.254.254' Method: INVITE
    -- Executing [s@macro-exten-vm:10] GotoIf("SIP/999-b5a03268", "0?exit,return") in new stack
    -- Executing [s@macro-exten-vm:11] Set("SIP/999-b5a03268", "SV_DIALSTATUS=NOANSWER") in new stack
    -- Executing [s@macro-exten-vm:12] GosubIf("SIP/999-b5a03268", "0?docfu,1") in new stack
    -- Executing [s@macro-exten-vm:13] GosubIf("SIP/999-b5a03268", "0?docfb,1") in new stack
    -- Executing [s@macro-exten-vm:14] Set("SIP/999-b5a03268", "DIALSTATUS=NOANSWER") in new stack
    -- Executing [s@macro-exten-vm:15] NoOp("SIP/999-b5a03268", "Voicemail is '995'") in new stack
    -- Executing [s@macro-exten-vm:16] GotoIf("SIP/999-b5a03268", "0?s-NOANSWER,1") in new stack
    -- Executing [s@macro-exten-vm:17] NoOp("SIP/999-b5a03268", "Sending to Voicemail box 995") in new stack
    -- Executing [s@macro-exten-vm:18] Macro("SIP/999-b5a03268", "vm,995,NOANSWER,") in new stack
    -- Executing [s@macro-vm:1] Macro("SIP/999-b5a03268", "user-callerid,SKIPTTL") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/999-b5a03268", "AMPUSER=999") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/999-b5a03268", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/999-b5a03268", "0?Set(REALCALLERIDNUM=999)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/999-b5a03268", "AMPUSER=999") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/999-b5a03268", "AMPUSERCIDNAME=Oda") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/999-b5a03268", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/999-b5a03268", "AMPUSERCID=999") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/999-b5a03268", "CALLERID(all)="Oda" <999>") in new stack
    -- Executing [s@macro-user-callerid:9] Set("SIP/999-b5a03268", "REALCALLERIDNUM=999") in new stack
    -- Executing [s@macro-user-callerid:10] ExecIf("SIP/999-b5a03268", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [s@macro-user-callerid:11] GotoIf("SIP/999-b5a03268", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,20)
    -- Executing [s@macro-user-callerid:20] NoOp("SIP/999-b5a03268", "Using CallerID "Oda" <999>") in new stack
    -- Executing [s@macro-vm:2] Set("SIP/999-b5a03268", "VMGAIN=""") in new stack
    -- Executing [s@macro-vm:3] GotoIf("SIP/999-b5a03268", "1?vmx,1") in new stack
    -- Goto (macro-vm,vmx,1)
    -- Executing [vmx@macro-vm:1] GotoIf("SIP/999-b5a03268", "0?s-NOANSWER,1") in new stack
    -- Executing [vmx@macro-vm:2] Set("SIP/999-b5a03268", "MODE=unavail") in new stack
    -- Executing [vmx@macro-vm:3] GotoIf("SIP/999-b5a03268", "1?notdirect") in new stack
    -- Goto (macro-vm,vmx,5)
    -- Executing [vmx@macro-vm:5] NoOp("SIP/999-b5a03268", "Checking if ext 995 is enabled: enabled") in new stack
    -- Executing [vmx@macro-vm:6] GotoIf("SIP/999-b5a03268", "0?s-NOANSWER,1") in new stack
    -- Executing [vmx@macro-vm:7] Macro("SIP/999-b5a03268", "get-vmcontext,995") in new stack
    -- Executing [s@macro-get-vmcontext:1] Set("SIP/999-b5a03268", "VMCONTEXT=default") in new stack
    -- Executing [s@macro-get-vmcontext:2] GotoIf("SIP/999-b5a03268", "0?200:300") in new stack
    -- Goto (macro-get-vmcontext,s,300)
    -- Executing [s@macro-get-vmcontext:300] NoOp("SIP/999-b5a03268", "") in new stack
    -- Executing [vmx@macro-vm:8] AGI("SIP/999-b5a03268", "checksound.agi,/var/spool/asterisk/voicemail/default/995/temp") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/checksound.agi
 checksound.agi,/var/spool/asterisk/voicemail/default/995/temp: VmX requires: /var/spool/asterisk/voicemail/default/995/temp.wav or .WAV exist in order to function
    -- <SIP/999-b5a03268>AGI Script checksound.agi completed, returning 0
    -- Executing [vmx@macro-vm:9] GotoIf("SIP/999-b5a03268", "0?tmpgreet") in new stack
    -- Executing [vmx@macro-vm:10] AGI("SIP/999-b5a03268", "checksound.agi,/var/spool/asterisk/voicemail/default/995/unavail") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/checksound.agi
    -- <SIP/999-b5a03268>AGI Script checksound.agi completed, returning 0
    -- Executing [vmx@macro-vm:11] GotoIf("SIP/999-b5a03268", "0?nofile") in new stack
    -- Executing [vmx@macro-vm:12] Set("SIP/999-b5a03268", "LOOPCOUNT=0") in new stack
    -- Executing [vmx@macro-vm:13] GotoIf("SIP/999-b5a03268", "1?vmxtime") in new stack
    -- Goto (macro-vm,vmx,15)
    -- Executing [vmx@macro-vm:15] GotoIf("SIP/999-b5a03268", "1?vmxloops") in new stack
    -- Goto (macro-vm,vmx,17)
    -- Executing [vmx@macro-vm:17] GotoIf("SIP/999-b5a03268", "1?vmxanswer") in new stack
    -- Goto (macro-vm,vmx,19)
    -- Executing [vmx@macro-vm:19] Answer("SIP/999-b5a03268", "") in new stack
Audio is at 192.168.254.254 port 33808
Adding codec 0x8 (alaw) to SDP
trixbox1*CLI> 
<--- Reliably Transmitting (no NAT) to 192.168.254.11:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.254.11:5060;branch=z9hG4bK-e34764ba;received=192.168.254.11
From: ilker <sip:999@192.168.254.254>;tag=a5b4e3ee2377931eo0
To: <sip:995@192.168.254.254>;tag=as1ba4eab0
Call-ID: 41fb902e-f8e6d3de@192.168.254.11
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:995@192.168.254.254>
Content-Type: application/sdp
Content-Length: 224

v=0
o=root 531870305 531870305 IN IP4 192.168.254.254
s=Asterisk PBX 1.6.0.10-FONCORE-r40
c=IN IP4 192.168.254.254
t=0 0
m=audio 33808 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
trixbox1*CLI> 
<--- SIP read from UDP://192.168.254.11:5060 --->
ACK sip:995@192.168.254.254 SIP/2.0
Via: SIP/2.0/UDP 192.168.254.11:5060;branch=z9hG4bK-6bc430f3
From: ilker <sip:999@192.168.254.254>;tag=a5b4e3ee2377931eo0
To: <sip:995@192.168.254.254>;tag=as1ba4eab0
Call-ID: 41fb902e-f8e6d3de@192.168.254.11
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username="999",realm="asterisk",nonce="501b1064",uri="sip:995@192.168.254.254",algorithm=MD5,response="09f395f4a4cd35a945427ee18e40ae5e"
Contact: ilker <sip:999@192.168.254.11:5060>
User-Agent: Linksys/PAP2-3.1.22(LS)
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
    -- Executing [vmx@macro-vm:20] Read("SIP/999-b5a03268", "ACTION,/var/spool/asterisk/voicemail/default/995/unavail,1,skip,1,2") in new stack
    -- Accepting a maximum of 1 digits.
    -- <SIP/999-b5a03268> Playing '/var/spool/asterisk/voicemail/default/995/unavail.slin' (language 'en')
    -- User entered nothing.
    -- Executing [vmx@macro-vm:21] GotoIf("SIP/999-b5a03268", "0?checkopt") in new stack
    -- Executing [vmx@macro-vm:22] NoOp("SIP/999-b5a03268", "Timeout: going to timeout dest") in new stack
    -- Executing [vmx@macro-vm:23] Set("SIP/999-b5a03268", "VMX_OPTS=") in new stack
    -- Executing [vmx@macro-vm:24] GotoIf("SIP/999-b5a03268", "0?chktime") in new stack
    -- Executing [vmx@macro-vm:25] Set("SIP/999-b5a03268", "VMX_OPTS=s") in new stack
    -- Executing [vmx@macro-vm:26] GotoIf("SIP/999-b5a03268", "1?dotime") in new stack
    -- Goto (macro-vm,vmx,32)
    -- Executing [vmx@macro-vm:32] Goto("SIP/999-b5a03268", ",dovm,1") in new stack
    -- Goto (macro-vm,dovm,1)
    -- Executing [dovm@macro-vm:1] NoOp("SIP/999-b5a03268", "VMX Timeout - go to voicemail") in new stack
    -- Executing [dovm@macro-vm:2] VoiceMail("SIP/999-b5a03268", "995@default,s") in new stack
    -- <SIP/999-b5a03268> Playing 'beep.gsm' (language 'en')
    -- Recording the message
    -- x=0, open writing:  /var/spool/asterisk/voicemail/default/995/tmp/NfpUu4 format: wav49, 0x84cea20
    -- x=1, open writing:  /var/spool/asterisk/voicemail/default/995/tmp/NfpUu4 format: wav, 0x8584ca0
trixbox1*CLI> 
<--- SIP read from UDP://192.168.254.11:5060 --->
BYE sip:995@192.168.254.254 SIP/2.0
Via: SIP/2.0/UDP 192.168.254.11:5060;branch=z9hG4bK-2bc2ae4f
From: ilker <sip:999@192.168.254.254>;tag=a5b4e3ee2377931eo0
To: <sip:995@192.168.254.254>;tag=as1ba4eab0
Call-ID: 41fb902e-f8e6d3de@192.168.254.11
CSeq: 103 BYE
Max-Forwards: 70
Authorization: Digest username="999",realm="asterisk",nonce="501b1064",uri="sip:995@192.168.254.254",algorithm=MD5,response="bdf96653b0d0c24276f29dca45fbccb9"
User-Agent: Linksys/PAP2-3.1.22(LS)
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.254.11 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.254.11:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.254.11:5060;branch=z9hG4bK-2bc2ae4f;received=192.168.254.11
From: ilker <sip:999@192.168.254.254>;tag=a5b4e3ee2377931eo0
To: <sip:995@192.168.254.254>;tag=as1ba4eab0
Call-ID: 41fb902e-f8e6d3de@192.168.254.11
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0


<------------>
    -- User hung up
Scheduling destruction of SIP dialog '51e50e80105bdfd7576b766e13a50b44@192.168.254.254' in 6400 ms (Method: NOTIFY)
Reliably Transmitting (no NAT) to 192.168.254.5:5060:
NOTIFY sip:995@192.168.254.5:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.254.254:5060;branch=z9hG4bK6bbd0f24;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.254.254>;tag=as69630a9d
To: <sip:995@192.168.254.5:5060>
Contact: <sip:Unknown@192.168.254.254>
Call-ID: 51e50e80105bdfd7576b766e13a50b44@192.168.254.254
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 91

Messages-Waiting: yes
Message-Account: sip:000@192.168.254.254
Voice-Message: 1/0 (0/0)

---
  == Spawn extension (macro-vm, dovm, 2) exited non-zero on 'SIP/999-b5a03268' in macro 'vm'
trixbox1*CLI> 
<--- SIP read from UDP://192.168.254.5:5060 --->
SIP/2.0 200 OK
To: <sip:995@192.168.254.5:5060>;tag=e8621188cecb1025i0
From: "Unknown" <sip:Unknown@192.168.254.254>;tag=as69630a9d
Call-ID: 51e50e80105bdfd7576b766e13a50b44@192.168.254.254
CSeq: 102 NOTIFY
Via: SIP/2.0/UDP 192.168.254.254:5060;branch=z9hG4bK6bbd0f24;rport=5060
Server: Linksys/SPA3000-3.1.10(GWd)
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '51e50e80105bdfd7576b766e13a50b44@192.168.254.254' Method: NOTIFY
trixbox1*CLI> 
  == Spawn extension (macro-exten-vm, s, 18) exited non-zero on 'SIP/999-b5a03268' in macro 'exten-vm'
  == Spawn extension (access, 995, 1) exited non-zero on 'SIP/999-b5a03268'
    -- Executing [h@access:1] Macro("SIP/999-b5a03268", "hangupcall,") in new stack
    -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/999-b5a03268", "w") in new stack
    -- Executing [s@macro-hangupcall:2] NoCDR("SIP/999-b5a03268", "") in new stack
    -- Executing [s@macro-hangupcall:3] GotoIf("SIP/999-b5a03268", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,6)
    -- Executing [s@macro-hangupcall:6] GotoIf("SIP/999-b5a03268", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] GotoIf("SIP/999-b5a03268", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,11)
    -- Executing [s@macro-hangupcall:11] Hangup("SIP/999-b5a03268", "") in new stack
  == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/999-b5a03268' in macro 'hangupcall'
  == Spawn extension (access, h, 1) exited non-zero on 'SIP/999-b5a03268'
  == End MixMonitor Recording SIP/999-b5a03268
Really destroying SIP dialog '41fb902e-f8e6d3de@192.168.254.11' Method: BYE
trixbox1*CLI> 
<--- SIP read from UDP://192.168.254.11:5060 --->
REGISTER sip:192.168.254.254 SIP/2.0
Via: SIP/2.0/UDP 192.168.254.11:5060;branch=z9hG4bK-56b8f846
From: ilker <sip:999@192.168.254.254>;tag=f6df4bd23a7909fco0
To: ilker <sip:999@192.168.254.254>
Call-ID: 950a5717-97116ed4@192.168.254.11
CSeq: 34252 REGISTER
Max-Forwards: 70
Authorization: Digest username="999",realm="asterisk",nonce="52f9e636",uri="sip:192.168.254.254",algorithm=MD5,response="7e63cd84bfff9bbfce6a3ffa54fbc4eb"
Contact: ilker <sip:999@192.168.254.11:5060>;expires=3600
User-Agent: Linksys/PAP2-3.1.22(LS)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces


<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.254.11 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.254.11:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.254.11:5060;branch=z9hG4bK-56b8f846;received=192.168.254.11
From: ilker <sip:999@192.168.254.254>;tag=f6df4bd23a7909fco0
To: ilker <sip:999@192.168.254.254>;tag=as6cc5796a
Call-ID: 950a5717-97116ed4@192.168.254.11
CSeq: 34252 REGISTER
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4a347c4e"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '950a5717-97116ed4@192.168.254.11' in 32000 ms (Method: REGISTER)
trixbox1*CLI> 
<--- SIP read from UDP://192.168.254.11:5060 --->
REGISTER sip:192.168.254.254 SIP/2.0
Via: SIP/2.0/UDP 192.168.254.11:5060;branch=z9hG4bK-ed9a3abf
From: ilker <sip:999@192.168.254.254>;tag=f6df4bd23a7909fco0
To: ilker <sip:999@192.168.254.254>
Call-ID: 950a5717-97116ed4@192.168.254.11
CSeq: 34253 REGISTER
Max-Forwards: 70
Authorization: Digest username="999",realm="asterisk",nonce="4a347c4e",uri="sip:192.168.254.254",algorithm=MD5,response="cbb3b08d9ada3f6cda336cba18e857e1"
Contact: ilker <sip:999@192.168.254.11:5060>;expires=3600
User-Agent: Linksys/PAP2-3.1.22(LS)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces


<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.254.11 : 5060 (no NAT)
Reliably Transmitting (no NAT) to 192.168.254.11:5060:
OPTIONS sip:999@192.168.254.11:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.254.254:5060;branch=z9hG4bK50601b87;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.254.254>;tag=as60fdbfa3
To: <sip:999@192.168.254.11:5060>
Contact: <sip:Unknown@192.168.254.254>
Call-ID: 7dedf5e55c92e4b42606bd2943697edd@192.168.254.254
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Date: Tue, 29 Sep 2009 16:10:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (no NAT) to 192.168.254.11:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.254.11:5060;branch=z9hG4bK-ed9a3abf;received=192.168.254.11
From: ilker <sip:999@192.168.254.254>;tag=f6df4bd23a7909fco0
To: ilker <sip:999@192.168.254.254>;tag=as6cc5796a
Call-ID: 950a5717-97116ed4@192.168.254.11
CSeq: 34253 REGISTER
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Expires: 600
Contact: <sip:999@192.168.254.11:5060>;expires=600
Date: Tue, 29 Sep 2009 16:10:48 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '950a5717-97116ed4@192.168.254.11' in 32000 ms (Method: REGISTER)
trixbox1*CLI> 
<--- SIP read from UDP://192.168.254.11:5060 --->
SIP/2.0 200 OK
To: <sip:999@192.168.254.11:5060>;tag=31df190ad9bcde24i0
From: "Unknown" <sip:Unknown@192.168.254.254>;tag=as60fdbfa3
Call-ID: 7dedf5e55c92e4b42606bd2943697edd@192.168.254.254
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.254.254:5060;branch=z9hG4bK50601b87
Server: Linksys/PAP2-3.1.22(LS)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces


<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '7dedf5e55c92e4b42606bd2943697edd@192.168.254.254' Method: OPTIONS
trixbox1*CLI> 
<--- SIP read from UDP://192.168.254.5:5060 --->
REGISTER sip:192.168.254.254 SIP/2.0
Via: SIP/2.0/UDP 192.168.254.5:5060;branch=z9hG4bK-f57cc671
From: Home <sip:995@192.168.254.254>;tag=dc825a2c5d4bfd1o0
To: Home <sip:995@192.168.254.254>
Call-ID: 81f7cf29-f85305d9@192.168.254.5
CSeq: 53881 REGISTER
Max-Forwards: 70
Authorization: Digest username="995",realm="asterisk",nonce="4803d234",uri="sip:192.168.254.254",algorithm=MD5,response="8058252e683f4e131d8c9ee3bf5fd8b9"
Contact: Home <sip:995@192.168.254.5:5060>;expires=3600
User-Agent: Linksys/SPA3000-3.1.10(GWd)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura


<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.254.5 : 5060 (no NAT)
trixbox1*CLI> 
<--- Transmitting (no NAT) to 192.168.254.5:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.254.5:5060;branch=z9hG4bK-f57cc671;received=192.168.254.5
From: Home <sip:995@192.168.254.254>;tag=dc825a2c5d4bfd1o0
To: Home <sip:995@192.168.254.254>;tag=as2f01cc4e
Call-ID: 81f7cf29-f85305d9@192.168.254.5
CSeq: 53881 REGISTER
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3577bf64"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '81f7cf29-f85305d9@192.168.254.5' in 32000 ms (Method: REGISTER)
trixbox1*CLI> 
<--- SIP read from UDP://192.168.254.5:5060 --->
REGISTER sip:192.168.254.254 SIP/2.0
Via: SIP/2.0/UDP 192.168.254.5:5060;branch=z9hG4bK-43843aaa
From: Home <sip:995@192.168.254.254>;tag=dc825a2c5d4bfd1o0
To: Home <sip:995@192.168.254.254>
Call-ID: 81f7cf29-f85305d9@192.168.254.5
CSeq: 53882 REGISTER
Max-Forwards: 70
Authorization: Digest username="995",realm="asterisk",nonce="3577bf64",uri="sip:192.168.254.254",algorithm=MD5,response="323f70ba43305a5f5e2ac5a8eb48a7b5"
Contact: Home <sip:995@192.168.254.5:5060>;expires=3600
User-Agent: Linksys/SPA3000-3.1.10(GWd)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura


<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.254.5 : 5060 (no NAT)
Reliably Transmitting (no NAT) to 192.168.254.5:5060:
OPTIONS sip:995@192.168.254.5:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.254.254:5060;branch=z9hG4bK3dffccfa;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.254.254>;tag=as18e8fadd
To: <sip:995@192.168.254.5:5060>
Contact: <sip:Unknown@192.168.254.254>
Call-ID: 273b936106ac71e30606cddd2db2889b@192.168.254.254
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Date: Tue, 29 Sep 2009 16:10:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0


---
trixbox1*CLI> 
<--- Transmitting (no NAT) to 192.168.254.5:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.254.5:5060;branch=z9hG4bK-43843aaa;received=192.168.254.5
From: Home <sip:995@192.168.254.254>;tag=dc825a2c5d4bfd1o0
To: Home <sip:995@192.168.254.254>;tag=as2f01cc4e
Call-ID: 81f7cf29-f85305d9@192.168.254.5
CSeq: 53882 REGISTER
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Expires: 600
Contact: <sip:995@192.168.254.5:5060>;expires=600
Date: Tue, 29 Sep 2009 16:10:48 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '81f7cf29-f85305d9@192.168.254.5' in 32000 ms (Method: REGISTER)
trixbox1*CLI> 
<--- SIP read from UDP://192.168.254.5:5060 --->
SIP/2.0 200 OK
To: <sip:995@192.168.254.5:5060>;tag=e8621188cecb1025i0
From: "Unknown" <sip:Unknown@192.168.254.254>;tag=as18e8fadd
Call-ID: 273b936106ac71e30606cddd2db2889b@192.168.254.254
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.254.254:5060;branch=z9hG4bK3dffccfa;rport=5060
Server: Linksys/SPA3000-3.1.10(GWd)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura


<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '273b936106ac71e30606cddd2db2889b@192.168.254.254' Method: OPTIONS
Really destroying SIP dialog '950a5717-97116ed4@192.168.254.11' Method: REGISTER
Really destroying SIP dialog '81f7cf29-f85305d9@192.168.254.5' Method: REGISTER
Reliably Transmitting (no NAT) to 192.168.254.11:5060:
OPTIONS sip:999@192.168.254.11:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.254.254:5060;branch=z9hG4bK39dae65d;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.254.254>;tag=as62fa100d
To: <sip:999@192.168.254.11:5060>
Contact: <sip:Unknown@192.168.254.254>
Call-ID: 2839211f0700edf160975da946357931@192.168.254.254
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Date: Tue, 29 Sep 2009 16:11:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0


---
trixbox1*CLI> 
<--- SIP read from UDP://192.168.254.11:5060 --->
SIP/2.0 200 OK
To: <sip:999@192.168.254.11:5060>;tag=31df190ad9bcde24i0
From: "Unknown" <sip:Unknown@192.168.254.254>;tag=as62fa100d
Call-ID: 2839211f0700edf160975da946357931@192.168.254.254
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.254.254:5060;branch=z9hG4bK39dae65d
Server: Linksys/PAP2-3.1.22(LS)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces


<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '2839211f0700edf160975da946357931@192.168.254.254' Method: OPTIONS
Reliably Transmitting (no NAT) to 192.168.254.5:5060:
OPTIONS sip:995@192.168.254.5:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.254.254:5060;branch=z9hG4bK2a3f03c9;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.254.254>;tag=as5829ae6a
To: <sip:995@192.168.254.5:5060>
Contact: <sip:Unknown@192.168.254.254>
Call-ID: 4c2c2e0d7ea2a5dc063a9a3647121f35@192.168.254.254
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Date: Tue, 29 Sep 2009 16:11:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0


---
trixbox1*CLI> 
<--- SIP read from UDP://192.168.254.5:5060 --->
SIP/2.0 200 OK
To: <sip:995@192.168.254.5:5060>;tag=e8621188cecb1025i0
From: "Unknown" <sip:Unknown@192.168.254.254>;tag=as5829ae6a
Call-ID: 4c2c2e0d7ea2a5dc063a9a3647121f35@192.168.254.254
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.254.254:5060;branch=z9hG4bK2a3f03c9;rport=5060
Server: Linksys/SPA3000-3.1.10(GWd)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura


<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '4c2c2e0d7ea2a5dc063a9a3647121f35@192.168.254.254' Method: OPTIONS
trixbox1*CLI>