
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '54bb31572ab3350f7f448afe67fe7e6c@10.10.10.7' Method: OPTIONS

<--- SIP read from UDP:10.10.10.13:5063 --->
SIP/2.0 200 OK
To: <sip:204@10.10.10.13:5063>;tag=47db739b438d60f6i3
From: "Unknown" <sip:Unknown@10.10.10.7>;tag=as3bbf78e7
Call-ID: 1496b7d33388fe732dce8fc020a488d5@10.10.10.7
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 10.10.10.7:5060;branch=z9hG4bK5d952416
Server: Linksys/SPA942-5.1.15(a)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces


<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '1496b7d33388fe732dce8fc020a488d5@10.10.10.7' Method: OPTIONS

<--- SIP read from UDP:216.58.0.51:5060 --->
OPTIONS sip:6477232824@173.35.236.158 SIP/2.0
Via: SIP/2.0/UDP 216.58.0.51:5060;branch=z9hG4bK69e57879;rport
From: "asterisk" <sip:asterisk@216.58.0.51>;tag=as5fe01a68
To: <sip:6477232824@173.35.236.158>
Contact: <sip:asterisk@216.58.0.51>
Call-ID: 42c0d0646cc1736253d1b0602889388b@216.58.0.51
CSeq: 102 OPTIONS
User-Agent: CIA.com PBX
Max-Forwards: 70
Date: Fri, 13 May 2011 23:23:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------->
--- (13 headers 0 lines) ---
Looking for 6477232824 in from-sip-external (domain 173.35.236.158)

<--- Transmitting (NAT) to 216.58.0.51:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.58.0.51:5060;branch=z9hG4bK69e57879;received=216.58.0.51;rport=5060
From: "asterisk" <sip:asterisk@216.58.0.51>;tag=as5fe01a68
To: <sip:6477232824@173.35.236.158>;tag=as0cc3b40d
Call-ID: 42c0d0646cc1736253d1b0602889388b@216.58.0.51
CSeq: 102 OPTIONS
Server: FPBX-2.8.1(1.6.2.15)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:173.35.236.158>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '42c0d0646cc1736253d1b0602889388b@216.58.0.51' in 32000 ms (Method: OPTIONS)
Reliably Transmitting (NAT) to 216.58.0.51:5060:
OPTIONS sip:216.58.0.51 SIP/2.0
Via: SIP/2.0/UDP 173.35.236.158:5060;branch=z9hG4bK176b858d;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@173.35.236.158>;tag=as2a355b0e
To: <sip:216.58.0.51>
Contact: <sip:Unknown@173.35.236.158>
Call-ID: 48ab674a76af33e362624bd06a898ff2@173.35.236.158
CSeq: 102 OPTIONS
User-Agent: FPBX-2.8.1(1.6.2.15)
Date: Fri, 13 May 2011 23:11:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:216.58.0.51:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 173.35.236.158:5060;branch=z9hG4bK176b858d;received=173.35.236.158;rport=5060
From: "Unknown" <sip:Unknown@173.35.236.158>;tag=as2a355b0e
To: <sip:216.58.0.51>;tag=as53d05492
Call-ID: 48ab674a76af33e362624bd06a898ff2@173.35.236.158
CSeq: 102 OPTIONS
User-Agent: CIA.com PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Accept: application/sdp
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '48ab674a76af33e362624bd06a898ff2@173.35.236.158' Method: OPTIONS
pbx*CLI> sip show registry
Host                           dnsmgr Username       Refresh State                Reg.Time
216.58.0.51:5060               N      6477232824         105 Registered           Fri, 13 May 2011 19:11:11
1 SIP registrations.
  == Manager 'admin' logged on from 127.0.0.1
  == Manager 'admin' logged off from 127.0.0.1
pbx*CLI>
Disconnected from Asterisk server
Executing last minute cleanups
[root@pbx ~]# asterisk -rvvvvvv
Asterisk 1.6.2.15, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
  == Parsing '/etc/asterisk/asterisk.conf':   == Found
Connected to Asterisk 1.6.2.15 currently running on pbx (pid = 2872)
Verbosity is at least 22
  == Manager 'admin' logged on from 127.0.0.1
  == Manager 'admin' logged off from 127.0.0.1

<--- SIP read from UDP:216.58.0.51:5060 --->
INVITE sip:6477232824@173.35.236.158 SIP/2.0
Via: SIP/2.0/UDP 216.58.0.51:5060;branch=z9hG4bK31720642;rport
From: "Unknown" <sip:4168853558@216.58.0.51>;tag=as24891138
To: <sip:6477232824@173.35.236.158>
Contact: <sip:4168853558@216.58.0.51>
Call-ID: 0588e6fc65b21fa231853add2bce5533@216.58.0.51
CSeq: 102 INVITE
User-Agent: CIA.com PBX
Max-Forwards: 70
Remote-Party-ID: "Unknown" <sip:4168853558@216.58.0.51>;privacy=off;screen=no
Date: Fri, 13 May 2011 23:25:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 11364 11364 IN IP4 216.58.0.51
s=session
c=IN IP4 216.58.0.51
t=0 0
m=audio 12706 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->
--- (15 headers 14 lines) ---
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Sending to 216.58.0.51 : 5060 (NAT)
Using INVITE request as basis request - 0588e6fc65b21fa231853add2bce5533@216.58.0.51
Found peer '6477232824' for '4168853558' from 216.58.0.51:5060
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 216.58.0.51:12706
Looking for 6477232824 in outbound_cia (domain 173.35.236.158)

<--- Reliably Transmitting (NAT) to 216.58.0.51:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 216.58.0.51:5060;branch=z9hG4bK31720642;received=216.58.0.51;rport=5060
From: "Unknown" <sip:4168853558@216.58.0.51>;tag=as24891138
To: <sip:6477232824@173.35.236.158>;tag=as5f4bda61
Call-ID: 0588e6fc65b21fa231853add2bce5533@216.58.0.51
CSeq: 102 INVITE
Server: FPBX-2.8.1(1.6.2.15)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '0588e6fc65b21fa231853add2bce5533@216.58.0.51' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:216.58.0.51:5060 --->
ACK sip:6477232824@173.35.236.158 SIP/2.0
Via: SIP/2.0/UDP 216.58.0.51:5060;branch=z9hG4bK31720642;rport
From: "Unknown" <sip:4168853558@216.58.0.51>;tag=as24891138
To: <sip:6477232824@173.35.236.158>;tag=as5f4bda61
Contact: <sip:4168853558@216.58.0.51>
Call-ID: 0588e6fc65b21fa231853add2bce5533@216.58.0.51
CSeq: 102 ACK
User-Agent: CIA.com PBX
Max-Forwards: 70
Remote-Party-ID: "Unknown" <sip:4168853558@216.58.0.51>;privacy=off;screen=no
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '0588e6fc65b21fa231853add2bce5533@216.58.0.51' Method: ACK
  == Manager 'admin' logged on from 127.0.0.1
Really destroying SIP dialog '00df9c986975a7190ddc066763c1cae0@216.58.0.51' Method: OPTIONS
Really destroying SIP dialog '4d93253230ee72c6545e39ba038aec0e@127.0.0.1' Method: REGISTER
  == Manager 'admin' logged off from 127.0.0.1
pbx*CLI>