Parsing /etc/asterisk/asterisk.conf
Parsing /etc/asterisk/extconfig.conf

[Mar 10 12:01:12]   == Parsing '/etc/asterisk/asterisk.conf': [Mar 10 12:01:12] Found

[Mar 10 12:01:12]   == Parsing '/etc/asterisk/extconfig.conf': [Mar 10 12:01:12] Found

[Mar 10 12:01:12] Asterisk 1.2.13, Copyright (C) 1999 - 2006 Digium, Inc. and others.

[Mar 10 12:01:12] Created by Mark Spencer <markster@digium.com>

[Mar 10 12:01:12] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.

[Mar 10 12:01:12] This is free software, with components licensed under the GNU General Public

[Mar 10 12:01:12] License version 2 and other licenses; you are welcome to redistribute it under

[Mar 10 12:01:12] certain conditions. Type 'show license' for details.

[Mar 10 12:01:12] =========================================================================

[Mar 10 12:01:12]   == Parsing '/etc/asterisk/logger.conf': Parsing /etc/asterisk/logger.conf
[Mar 10 12:01:12] Found

[Mar 10 12:01:12] Asterisk Event Logger Started /var/log/asterisk/event_log

[Mar 10 12:01:12]   == Parsing '/etc/asterisk/dnsmgr.conf': [Mar 10 12:01:12] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/dnsmgr.conf
[Mar 10 12:01:12] Found

[Mar 10 12:01:12] Asterisk Dynamic Loader loading preload modules:

[Mar 10 12:01:12]   == Parsing '/etc/asterisk/modules.conf': [Mar 10 12:01:12] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/modules.conf
[Mar 10 12:01:12] Found

[Mar 10 12:01:12]   == Manager registered action Ping

[Mar 10 12:01:12]   == Manager registered action Events

[Mar 10 12:01:12]   == Manager registered action Logoff

[Mar 10 12:01:12]   == Manager registered action Hangup

[Mar 10 12:01:12]   == Manager registered action Status

[Mar 10 12:01:12]   == Manager registered action Setvar

[Mar 10 12:01:12]   == Manager registered action Getvar

[Mar 10 12:01:12]   == Manager registered action Redirect

[Mar 10 12:01:12]   == Manager registered action Originate

[Mar 10 12:01:12]   == Manager registered action Command

[Mar 10 12:01:12]   == Manager registered action ExtensionState

[Mar 10 12:01:12]   == Manager registered action AbsoluteTimeout

[Mar 10 12:01:12]   == Manager registered action MailboxStatus

[Mar 10 12:01:12]   == Manager registered action MailboxCount

[Mar 10 12:01:12]   == Manager registered action ListCommands

[Mar 10 12:01:12]   == Parsing '/etc/asterisk/manager.conf': [Mar 10 12:01:12] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/manager.conf
[Mar 10 12:01:12] Found

[Mar 10 12:01:12] Asterisk Management interface listening on port 5038

[Mar 10 12:01:12]   == Parsing '/etc/asterisk/cdr.conf': [Mar 10 12:01:12] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/cdr.conf
[Mar 10 12:01:12] Found
[Mar 10 12:01:12] NOTICE[15131]: cdr.c:1192 do_reload: CDR simple logging enabled.

[Mar 10 12:01:12]   == Parsing '/etc/asterisk/rtp.conf': [Mar 10 12:01:12] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/rtp.conf
[Mar 10 12:01:12] Found

[Mar 10 12:01:12]   == RTP Allocating from port range 10000 -> 20000

[Mar 10 12:01:12] Asterisk PBX Core Initializing

[Mar 10 12:01:12] Registering builtin applications:

[Mar 10 12:01:12]  [AbsoluteTimeout]

[Mar 10 12:01:12]   == Registered application 'AbsoluteTimeout'

[Mar 10 12:01:12]  [Answer]

[Mar 10 12:01:12]   == Registered application 'Answer'

[Mar 10 12:01:12]  [BackGround]

[Mar 10 12:01:12]   == Registered application 'BackGround'

[Mar 10 12:01:12]  [Busy]

[Mar 10 12:01:12]   == Registered application 'Busy'

[Mar 10 12:01:12]  [Congestion]

[Mar 10 12:01:12]   == Registered application 'Congestion'

[Mar 10 12:01:12]  [DigitTimeout]

[Mar 10 12:01:12]   == Registered application 'DigitTimeout'

[Mar 10 12:01:12]  [Goto]

[Mar 10 12:01:12]   == Registered application 'Goto'

[Mar 10 12:01:12]  [GotoIf]

[Mar 10 12:01:12]   == Registered application 'GotoIf'

[Mar 10 12:01:12]  [GotoIfTime]

[Mar 10 12:01:12]   == Registered application 'GotoIfTime'

[Mar 10 12:01:12]  [ExecIfTime]

[Mar 10 12:01:12]   == Registered application 'ExecIfTime'

[Mar 10 12:01:12]  [Hangup]

[Mar 10 12:01:12]   == Registered application 'Hangup'

[Mar 10 12:01:12]  [NoOp]

[Mar 10 12:01:12]   == Registered application 'NoOp'

[Mar 10 12:01:12]  [Progress]

[Mar 10 12:01:12]   == Registered application 'Progress'

[Mar 10 12:01:12]  [ResetCDR]

[Mar 10 12:01:12]   == Registered application 'ResetCDR'

[Mar 10 12:01:12]  [ResponseTimeout]

[Mar 10 12:01:12]   == Registered application 'ResponseTimeout'

[Mar 10 12:01:12]  [Ringing]

[Mar 10 12:01:12]   == Registered application 'Ringing'

[Mar 10 12:01:12]  [SayNumber]

[Mar 10 12:01:12]   == Registered application 'SayNumber'

[Mar 10 12:01:12]  [SayDigits]

[Mar 10 12:01:12]   == Registered application 'SayDigits'

[Mar 10 12:01:12]  [SayAlpha]

[Mar 10 12:01:12]   == Registered application 'SayAlpha'

[Mar 10 12:01:12]  [SayPhonetic]

[Mar 10 12:01:12]   == Registered application 'SayPhonetic'

[Mar 10 12:01:12]  [SetAccount]

[Mar 10 12:01:12]   == Registered application 'SetAccount'

[Mar 10 12:01:12]  [SetAMAFlags]

[Mar 10 12:01:12]   == Registered application 'SetAMAFlags'

[Mar 10 12:01:12]  [SetGlobalVar]

[Mar 10 12:01:12]   == Registered application 'SetGlobalVar'

[Mar 10 12:01:12]  [SetLanguage]

[Mar 10 12:01:12]   == Registered application 'SetLanguage'

[Mar 10 12:01:12]  [Set]

[Mar 10 12:01:12]   == Registered application 'Set'

[Mar 10 12:01:12]  [SetVar]

[Mar 10 12:01:12]   == Registered application 'SetVar'

[Mar 10 12:01:12]  [ImportVar]

[Mar 10 12:01:12]   == Registered application 'ImportVar'

[Mar 10 12:01:12]  [Wait]

[Mar 10 12:01:12]   == Registered application 'Wait'

[Mar 10 12:01:12]  [WaitExten]

[Mar 10 12:01:12]   == Registered application 'WaitExten'

[Mar 10 12:01:12]   == Manager registered action DBGet

[Mar 10 12:01:12]   == Manager registered action DBPut

[Mar 10 12:01:12]   == Parsing '/etc/asterisk/enum.conf': [Mar 10 12:01:12] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/enum.conf
[Mar 10 12:01:12] Found

[Mar 10 12:01:12] Asterisk Dynamic Loader Starting:

[Mar 10 12:01:12]   == Parsing '/etc/asterisk/modules.conf': [Mar 10 12:01:12] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/modules.conf
[Mar 10 12:01:12] Found

[Mar 10 12:01:12]  [res_musiconhold.so][Mar 10 12:01:12]  => (Music On Hold Resource)

[Mar 10 12:01:12]   == Registered application 'MusicOnHold'

[Mar 10 12:01:12]   == Registered application 'WaitMusicOnHold'

[Mar 10 12:01:12]   == Registered application 'SetMusicOnHold'

[Mar 10 12:01:12]   == Registered application 'StartMusicOnHold'

[Mar 10 12:01:12]   == Registered application 'StopMusicOnHold'

[Mar 10 12:01:12]   == Parsing '/etc/asterisk/musiconhold.conf': [Mar 10 12:01:12] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/musiconhold.conf
[Mar 10 12:01:12] Found
[Mar 10 12:01:12] WARNING[15131]: res_musiconhold.c:1017 load_moh_classes: The old musiconhold.conf syntax has been deprecated!  Please refer to the sample configuration for information on the new syntax.

[Mar 10 12:01:12]  [res_monitor.so][Mar 10 12:01:12]  => (Call Monitoring Resource)

[Mar 10 12:01:12]   == Registered application 'Monitor'

[Mar 10 12:01:12]   == Registered application 'StopMonitor'

[Mar 10 12:01:12]   == Registered application 'ChangeMonitor'

[Mar 10 12:01:12]   == Manager registered action Monitor

[Mar 10 12:01:12]   == Manager registered action StopMonitor

[Mar 10 12:01:12]   == Manager registered action ChangeMonitor

[Mar 10 12:01:12]  [res_indications.so][Mar 10 12:01:12]  => (Indications Configuration)

[Mar 10 12:01:12]   == Parsing '/etc/asterisk/indications.conf': [Mar 10 12:01:12] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/indications.conf
[Mar 10 12:01:12] Found

[Mar 10 12:01:12]     -- Registered indication country 'cl'

[Mar 10 12:01:12]     -- Registered indication country 'tw'

[Mar 10 12:01:12]     -- Registered indication country 'us'

[Mar 10 12:01:12]     -- Registered indication country 'au'

[Mar 10 12:01:12]     -- Registered indication country 'fr'

[Mar 10 12:01:12]     -- Registered indication country 'de'

[Mar 10 12:01:12]     -- Registered indication country 'nl'

[Mar 10 12:01:12]     -- Registered indication country 'uk'

[Mar 10 12:01:12]     -- Registered indication country 'fi'

[Mar 10 12:01:12]     -- Registered indication country 'no'

[Mar 10 12:01:12]     -- Registered indication country 'br'

[Mar 10 12:01:12]     -- Registered indication country 'za'

[Mar 10 12:01:12]     -- Registered indication country 'it'

[Mar 10 12:01:12]     -- Registered indication country 'us-o'

[Mar 10 12:01:12]     -- Registered indication country 'gr'

[Mar 10 12:01:12]     -- Registered indication country 'ru'

[Mar 10 12:01:12]     -- Registered indication country 'nz'

[Mar 10 12:01:12]     -- Setting default indication country to 'us'

[Mar 10 12:01:12]   == Registered application 'PlayTones'

[Mar 10 12:01:12]   == Registered application 'StopPlayTones'

[Mar 10 12:01:12]  [res_agi.so][Mar 10 12:01:12]  => (Asterisk Gateway Interface (AGI))

[Mar 10 12:01:12]   == Registered application 'DeadAGI'

[Mar 10 12:01:12]   == Registered application 'EAGI'

[Mar 10 12:01:12]   == Registered application 'AGI'

[Mar 10 12:01:12]  [res_features.so][Mar 10 12:01:12]  => (Call Features Resource)

[Mar 10 12:01:12]   == Parsing '/etc/asterisk/features.conf': [Mar 10 12:01:12] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/features.conf
[Mar 10 12:01:12] Found
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:3674 ast_context_create: Registered context 'parkedcalls'

[Mar 10 12:01:12]     -- Registered extension context 'parkedcalls'
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '700' priority 1 to parkedcalls

[Mar 10 12:01:12]     -- Added extension '700' priority 1 to parkedcalls

[Mar 10 12:01:12]   == Registered application 'ParkedCall'

[Mar 10 12:01:12]   == Registered application 'Park'

[Mar 10 12:01:12]   == Manager registered action ParkedCalls

[Mar 10 12:01:12]  [res_crypto.so][Mar 10 12:01:12]  => (Cryptographic Digital Signatures)

[Mar 10 12:01:12]     -- Loaded PUBLIC key 'freeworlddialup'
[Mar 10 12:01:12] DEBUG[15131]: res_crypto.c:257 try_load_key: Key 'freeworlddialup' loaded OK

[Mar 10 12:01:12]     -- Loaded PUBLIC key 'iaxtel'
[Mar 10 12:01:12] DEBUG[15131]: res_crypto.c:257 try_load_key: Key 'iaxtel' loaded OK

[Mar 10 12:01:12]  [res_adsi.so][Mar 10 12:01:12]  => (ADSI Resource)

[Mar 10 12:01:12]   == Parsing '/etc/asterisk/adsi.conf': [Mar 10 12:01:12] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/adsi.conf
[Mar 10 12:01:12] Found

[Mar 10 12:01:12]  [pbx_spool.so][Mar 10 12:01:12]  => (Outgoing Spool Support)

[Mar 10 12:01:12]  [pbx_ael.so][Mar 10 12:01:12]  => (Asterisk Extension Language Compiler)
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:3674 ast_context_create: Registered context 'macro-std-exten-ael'

[Mar 10 12:01:12]     -- Registered extension context 'macro-std-exten-ael'
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension 's' priority 1 to macro-std-exten-ael

[Mar 10 12:01:12]     -- Added extension 's' priority 1 to macro-std-exten-ael
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 's' priority 2 to macro-std-exten-ael

[Mar 10 12:01:12]     -- Added extension 's' priority 2 to macro-std-exten-ael
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 's' priority 3 to macro-std-exten-ael

[Mar 10 12:01:12]     -- Added extension 's' priority 3 to macro-std-exten-ael
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 's' priority 4 to macro-std-exten-ael

[Mar 10 12:01:12]     -- Added extension 's' priority 4 to macro-std-exten-ael
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 's' priority 5 to macro-std-exten-ael

[Mar 10 12:01:12]     -- Added extension 's' priority 5 to macro-std-exten-ael
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension 'sw-4-BUSY' priority 1 to macro-std-exten-ael

[Mar 10 12:01:12]     -- Added extension 'sw-4-BUSY' priority 1 to macro-std-exten-ael
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'sw-4-BUSY' priority 2 to macro-std-exten-ael

[Mar 10 12:01:12]     -- Added extension 'sw-4-BUSY' priority 2 to macro-std-exten-ael
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_sw-4-.' priority 1 to macro-std-exten-ael

[Mar 10 12:01:12]     -- Added extension '_sw-4-.' priority 1 to macro-std-exten-ael
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension 'a' priority 1 to macro-std-exten-ael

[Mar 10 12:01:12]     -- Added extension 'a' priority 1 to macro-std-exten-ael
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'a' priority 2 to macro-std-exten-ael

[Mar 10 12:01:12]     -- Added extension 'a' priority 2 to macro-std-exten-ael
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:3674 ast_context_create: Registered context 'ael-demo'

[Mar 10 12:01:12]     -- Registered extension context 'ael-demo'
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension 's' priority 1 to ael-demo

[Mar 10 12:01:12]     -- Added extension 's' priority 1 to ael-demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 's' priority 2 to ael-demo

[Mar 10 12:01:12]     -- Added extension 's' priority 2 to ael-demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 's' priority 3 to ael-demo

[Mar 10 12:01:12]     -- Added extension 's' priority 3 to ael-demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 's' priority 4 to ael-demo

[Mar 10 12:01:12]     -- Added extension 's' priority 4 to ael-demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 's' priority 5 to ael-demo

[Mar 10 12:01:12]     -- Added extension 's' priority 5 to ael-demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 's' priority 6 to ael-demo

[Mar 10 12:01:12]     -- Added extension 's' priority 6 to ael-demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 's' priority 8 to ael-demo

[Mar 10 12:01:12]     -- Added extension 's' priority 8 to ael-demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 's' priority 9 to ael-demo

[Mar 10 12:01:12]     -- Added extension 's' priority 9 to ael-demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 's' priority 10 to ael-demo

[Mar 10 12:01:12]     -- Added extension 's' priority 10 to ael-demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 's' priority 11 to ael-demo

[Mar 10 12:01:12]     -- Added extension 's' priority 11 to ael-demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 's' priority 12 to ael-demo

[Mar 10 12:01:12]     -- Added extension 's' priority 12 to ael-demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4828 ast_add_extension2: Added extension 's' priority 7 to ael-demo

[Mar 10 12:01:12]     -- Added extension 's' priority 7 to ael-demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '2' priority 1 to ael-demo

[Mar 10 12:01:12]     -- Added extension '2' priority 1 to ael-demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '2' priority 2 to ael-demo

[Mar 10 12:01:12]     -- Added extension '2' priority 2 to ael-demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '3' priority 1 to ael-demo

[Mar 10 12:01:12]     -- Added extension '3' priority 1 to ael-demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '3' priority 2 to ael-demo

[Mar 10 12:01:12]     -- Added extension '3' priority 2 to ael-demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '500' priority 1 to ael-demo

[Mar 10 12:01:12]     -- Added extension '500' priority 1 to ael-demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '500' priority 2 to ael-demo

[Mar 10 12:01:12]     -- Added extension '500' priority 2 to ael-demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '500' priority 3 to ael-demo

[Mar 10 12:01:12]     -- Added extension '500' priority 3 to ael-demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '500' priority 4 to ael-demo

[Mar 10 12:01:12]     -- Added extension '500' priority 4 to ael-demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '600' priority 1 to ael-demo

[Mar 10 12:01:12]     -- Added extension '600' priority 1 to ael-demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '600' priority 2 to ael-demo

[Mar 10 12:01:12]     -- Added extension '600' priority 2 to ael-demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '600' priority 3 to ael-demo

[Mar 10 12:01:12]     -- Added extension '600' priority 3 to ael-demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '600' priority 4 to ael-demo

[Mar 10 12:01:12]     -- Added extension '600' priority 4 to ael-demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_1234' priority 1 to ael-demo

[Mar 10 12:01:12]     -- Added extension '_1234' priority 1 to ael-demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '#' priority 1 to ael-demo

[Mar 10 12:01:12]     -- Added extension '#' priority 1 to ael-demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '#' priority 2 to ael-demo

[Mar 10 12:01:12]     -- Added extension '#' priority 2 to ael-demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension 't' priority 1 to ael-demo

[Mar 10 12:01:12]     -- Added extension 't' priority 1 to ael-demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension 'i' priority 1 to ael-demo

[Mar 10 12:01:12]     -- Added extension 'i' priority 1 to ael-demo

[Mar 10 12:01:12]  [pbx_loopback.so][Mar 10 12:01:12]  => (Loopback Switch)

[Mar 10 12:01:12]  [pbx_dundi.so][Mar 10 12:01:12]  => (Distributed Universal Number Discovery (DUNDi))

[Mar 10 12:01:12]   == Parsing '/etc/asterisk/dundi.conf': [Mar 10 12:01:12] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/dundi.conf
[Mar 10 12:01:12] Found
[Mar 10 12:01:12] DEBUG[15131]: pbx_dundi.c:422 reset_global_eid: Seeding global EID '00:01:03:dc:49:ec' from 'eth0'

[Mar 10 12:01:12]   == Using TOS bits 0

[Mar 10 12:01:12]   == DUNDi Ready and Listening on 0.0.0.0 port 4520

[Mar 10 12:01:12]   == Registered application 'DUNDiLookup'

[Mar 10 12:01:12]   == Registered custom function DUNDILOOKUP

[Mar 10 12:01:12]  [pbx_config.so][Mar 10 12:01:12]  => (Text Extension Configuration)

[Mar 10 12:01:12]   == Parsing '/etc/asterisk/extensions.conf': [Mar 10 12:01:12] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/extensions.conf
[Mar 10 12:01:12] Found

[Mar 10 12:01:12]   == Parsing '/etc/asterisk/express.demonstration.extensions.meetme.conf': [Mar 10 12:01:12] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.meetme.conf
[Mar 10 12:01:12] Found

[Mar 10 12:01:12]   == Parsing '/etc/asterisk/express.demonstration.dnis.conf': [Mar 10 12:01:12] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/express.demonstration.dnis.conf
[Mar 10 12:01:12] Found

[Mar 10 12:01:12]   == Parsing '/etc/asterisk/express.demonstration.extensions.meetme.conf': [Mar 10 12:01:12] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.meetme.conf
[Mar 10 12:01:12] Found

[Mar 10 12:01:12]   == Setting global variable 'CONSOLE' to 'Console/dsp'

[Mar 10 12:01:12]   == Setting global variable 'IAXINFO' to 'guest'

[Mar 10 12:01:12]   == Setting global variable 'TRUNK' to 'Zap/g2'

[Mar 10 12:01:12]   == Setting global variable 'TRUNKMSD' to '1'

[Mar 10 12:01:12]   == Setting global variable 'DIAL_TIMEOUT' to '20'

[Mar 10 12:01:12]   == Setting global variable 'SMVOICE_DIAL_TIMEOUT' to '60'

[Mar 10 12:01:12]   == Setting global variable 'SMVOICE_ANNOUNCE_CALLER' to '1'

[Mar 10 12:01:12]   == Setting global variable 'SMVOICE_DIAL_LONG_TIMEOUT' to '120'

[Mar 10 12:01:12]   == Setting global variable 'OPERATOR' to '510'

[Mar 10 12:01:12]   == Setting global variable 'OPERATOR_TECHNOLOGY' to 'SIP'

[Mar 10 12:01:12]   == Setting global variable 'SUPPORT' to '216'

[Mar 10 12:01:12]   == Setting global variable 'SALES' to '217'

[Mar 10 12:01:12]   == Setting global variable 'INTERCOM' to 'Zap/8'

[Mar 10 12:01:12]   == Setting global variable 'SMVOICE_ONHOLD' to ''
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:3674 ast_context_create: Registered context 'iaxtel700'

[Mar 10 12:01:12]     -- Registered extension context 'iaxtel700'
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_91700XXXXXXX' priority 1 to iaxtel700

[Mar 10 12:01:12]     -- Added extension '_91700XXXXXXX' priority 1 to iaxtel700
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:3674 ast_context_create: Registered context 'iaxprovider'

[Mar 10 12:01:12]     -- Registered extension context 'iaxprovider'
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:3674 ast_context_create: Registered context 'trunkint'

[Mar 10 12:01:12]     -- Registered extension context 'trunkint'
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_9011.' priority 1 to trunkint

[Mar 10 12:01:12]     -- Added extension '_9011.' priority 1 to trunkint
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_9011.' priority 2 to trunkint

[Mar 10 12:01:12]     -- Added extension '_9011.' priority 2 to trunkint
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:3674 ast_context_create: Registered context 'trunkld'

[Mar 10 12:01:12]     -- Registered extension context 'trunkld'
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_91NXXNXXXXXX' priority 1 to trunkld

[Mar 10 12:01:12]     -- Added extension '_91NXXNXXXXXX' priority 1 to trunkld
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_91NXXNXXXXXX' priority 2 to trunkld

[Mar 10 12:01:12]     -- Added extension '_91NXXNXXXXXX' priority 2 to trunkld
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:3674 ast_context_create: Registered context 'trunklocal'

[Mar 10 12:01:12]     -- Registered extension context 'trunklocal'
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_9NXXXXXX' priority 1 to trunklocal

[Mar 10 12:01:12]     -- Added extension '_9NXXXXXX' priority 1 to trunklocal
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_9NXXXXXX' priority 2 to trunklocal

[Mar 10 12:01:12]     -- Added extension '_9NXXXXXX' priority 2 to trunklocal
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:3674 ast_context_create: Registered context 'trunktollfree'

[Mar 10 12:01:12]     -- Registered extension context 'trunktollfree'
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_91800NXXXXXX' priority 1 to trunktollfree

[Mar 10 12:01:12]     -- Added extension '_91800NXXXXXX' priority 1 to trunktollfree
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_91800NXXXXXX' priority 2 to trunktollfree

[Mar 10 12:01:12]     -- Added extension '_91800NXXXXXX' priority 2 to trunktollfree
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_91888NXXXXXX' priority 1 to trunktollfree

[Mar 10 12:01:12]     -- Added extension '_91888NXXXXXX' priority 1 to trunktollfree
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_91888NXXXXXX' priority 2 to trunktollfree

[Mar 10 12:01:12]     -- Added extension '_91888NXXXXXX' priority 2 to trunktollfree
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '_91877NXXXXXX' priority 1 to trunktollfree

[Mar 10 12:01:12]     -- Added extension '_91877NXXXXXX' priority 1 to trunktollfree
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_91877NXXXXXX' priority 2 to trunktollfree

[Mar 10 12:01:12]     -- Added extension '_91877NXXXXXX' priority 2 to trunktollfree
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '_91866NXXXXXX' priority 1 to trunktollfree

[Mar 10 12:01:12]     -- Added extension '_91866NXXXXXX' priority 1 to trunktollfree
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_91866NXXXXXX' priority 2 to trunktollfree

[Mar 10 12:01:12]     -- Added extension '_91866NXXXXXX' priority 2 to trunktollfree
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:3674 ast_context_create: Registered context 'international'

[Mar 10 12:01:12]     -- Registered extension context 'international'

[Mar 10 12:01:12]     -- Including context 'longdistance' in context 'international'

[Mar 10 12:01:12]     -- Including context 'trunkint' in context 'international'
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:3674 ast_context_create: Registered context 'longdistance'

[Mar 10 12:01:12]     -- Registered extension context 'longdistance'

[Mar 10 12:01:12]     -- Including context 'local' in context 'longdistance'

[Mar 10 12:01:12]     -- Including context 'trunkld' in context 'longdistance'
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:3674 ast_context_create: Registered context 'local'

[Mar 10 12:01:12]     -- Registered extension context 'local'

[Mar 10 12:01:12]     -- Including context 'default' in context 'local'

[Mar 10 12:01:12]     -- Including context 'parkedcalls' in context 'local'

[Mar 10 12:01:12]     -- Including context 'trunklocal' in context 'local'

[Mar 10 12:01:12]     -- Including context 'iaxtel700' in context 'local'

[Mar 10 12:01:12]     -- Including context 'trunktollfree' in context 'local'

[Mar 10 12:01:12]     -- Including context 'iaxprovider' in context 'local'
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:3674 ast_context_create: Registered context 'macro-stdexten'

[Mar 10 12:01:12]     -- Registered extension context 'macro-stdexten'
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension 's' priority 1 to macro-stdexten

[Mar 10 12:01:12]     -- Added extension 's' priority 1 to macro-stdexten
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 's' priority 2 to macro-stdexten

[Mar 10 12:01:12]     -- Added extension 's' priority 2 to macro-stdexten
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension 's-NOANSWER' priority 1 to macro-stdexten

[Mar 10 12:01:12]     -- Added extension 's-NOANSWER' priority 1 to macro-stdexten
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 's-NOANSWER' priority 2 to macro-stdexten

[Mar 10 12:01:12]     -- Added extension 's-NOANSWER' priority 2 to macro-stdexten
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension 's-BUSY' priority 1 to macro-stdexten

[Mar 10 12:01:12]     -- Added extension 's-BUSY' priority 1 to macro-stdexten
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 's-BUSY' priority 2 to macro-stdexten

[Mar 10 12:01:12]     -- Added extension 's-BUSY' priority 2 to macro-stdexten
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_s-.' priority 1 to macro-stdexten

[Mar 10 12:01:12]     -- Added extension '_s-.' priority 1 to macro-stdexten
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension 'a' priority 1 to macro-stdexten

[Mar 10 12:01:12]     -- Added extension 'a' priority 1 to macro-stdexten
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:3674 ast_context_create: Registered context 'demo'

[Mar 10 12:01:12]     -- Registered extension context 'demo'
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension 's' priority 1 to demo

[Mar 10 12:01:12]     -- Added extension 's' priority 1 to demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 's' priority 2 to demo

[Mar 10 12:01:12]     -- Added extension 's' priority 2 to demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 's' priority 3 to demo

[Mar 10 12:01:12]     -- Added extension 's' priority 3 to demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 's' priority 4 to demo

[Mar 10 12:01:12]     -- Added extension 's' priority 4 to demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 's' priority 5 to demo

[Mar 10 12:01:12]     -- Added extension 's' priority 5 to demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 's' priority 6 to demo

[Mar 10 12:01:12]     -- Added extension 's' priority 6 to demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '2' priority 1 to demo

[Mar 10 12:01:12]     -- Added extension '2' priority 1 to demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '2' priority 2 to demo

[Mar 10 12:01:12]     -- Added extension '2' priority 2 to demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '3' priority 1 to demo

[Mar 10 12:01:12]     -- Added extension '3' priority 1 to demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '3' priority 2 to demo

[Mar 10 12:01:12]     -- Added extension '3' priority 2 to demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '1000' priority 1 to demo

[Mar 10 12:01:12]     -- Added extension '1000' priority 1 to demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '1234' priority 1 to demo

[Mar 10 12:01:12]     -- Added extension '1234' priority 1 to demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '1234' priority 2 to demo

[Mar 10 12:01:12]     -- Added extension '1234' priority 2 to demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '1235' priority 1 to demo

[Mar 10 12:01:12]     -- Added extension '1235' priority 1 to demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '1236' priority 1 to demo

[Mar 10 12:01:12]     -- Added extension '1236' priority 1 to demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '1236' priority 2 to demo

[Mar 10 12:01:12]     -- Added extension '1236' priority 2 to demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '#' priority 1 to demo

[Mar 10 12:01:12]     -- Added extension '#' priority 1 to demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '#' priority 2 to demo

[Mar 10 12:01:12]     -- Added extension '#' priority 2 to demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension 't' priority 1 to demo

[Mar 10 12:01:12]     -- Added extension 't' priority 1 to demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension 'i' priority 1 to demo

[Mar 10 12:01:12]     -- Added extension 'i' priority 1 to demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '500' priority 1 to demo

[Mar 10 12:01:12]     -- Added extension '500' priority 1 to demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '500' priority 2 to demo

[Mar 10 12:01:12]     -- Added extension '500' priority 2 to demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '500' priority 3 to demo

[Mar 10 12:01:12]     -- Added extension '500' priority 3 to demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '500' priority 4 to demo

[Mar 10 12:01:12]     -- Added extension '500' priority 4 to demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '600' priority 1 to demo

[Mar 10 12:01:12]     -- Added extension '600' priority 1 to demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '600' priority 2 to demo

[Mar 10 12:01:12]     -- Added extension '600' priority 2 to demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '600' priority 3 to demo

[Mar 10 12:01:12]     -- Added extension '600' priority 3 to demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '600' priority 4 to demo

[Mar 10 12:01:12]     -- Added extension '600' priority 4 to demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '8500' priority 1 to demo

[Mar 10 12:01:12]     -- Added extension '8500' priority 1 to demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '8500' priority 2 to demo

[Mar 10 12:01:12]     -- Added extension '8500' priority 2 to demo
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:3674 ast_context_create: Registered context 'default'

[Mar 10 12:01:12]     -- Registered extension context 'default'
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension 's' priority 1 to default

[Mar 10 12:01:12]     -- Added extension 's' priority 1 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 's' priority 2 to default

[Mar 10 12:01:12]     -- Added extension 's' priority 2 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 's' priority 3 to default

[Mar 10 12:01:12]     -- Added extension 's' priority 3 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 's' priority 4 to default

[Mar 10 12:01:12]     -- Added extension 's' priority 4 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 's' priority 5 to default

[Mar 10 12:01:12]     -- Added extension 's' priority 5 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 's' priority 6 to default

[Mar 10 12:01:12]     -- Added extension 's' priority 6 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 's' priority 7 to default

[Mar 10 12:01:12]     -- Added extension 's' priority 7 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 's' priority 8 to default

[Mar 10 12:01:12]     -- Added extension 's' priority 8 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 's' priority 9 to default

[Mar 10 12:01:12]     -- Added extension 's' priority 9 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '3173241051' priority 1 to default

[Mar 10 12:01:12]     -- Added extension '3173241051' priority 1 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '3173241052' priority 1 to default

[Mar 10 12:01:12]     -- Added extension '3173241052' priority 1 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension 'operator' priority 1 to default

[Mar 10 12:01:12]     -- Added extension 'operator' priority 1 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'operator' priority 2 to default

[Mar 10 12:01:12]     -- Added extension 'operator' priority 2 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'operator' priority 3 to default

[Mar 10 12:01:12]     -- Added extension 'operator' priority 3 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'operator' priority 4 to default

[Mar 10 12:01:12]     -- Added extension 'operator' priority 4 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'operator' priority 5 to default

[Mar 10 12:01:12]     -- Added extension 'operator' priority 5 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'operator' priority 6 to default

[Mar 10 12:01:12]     -- Added extension 'operator' priority 6 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension 'operator-CHANUNAVAIL' priority 1 to default

[Mar 10 12:01:12]     -- Added extension 'operator-CHANUNAVAIL' priority 1 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension 'operator-CONGESTION' priority 1 to default

[Mar 10 12:01:12]     -- Added extension 'operator-CONGESTION' priority 1 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension 'operator-NOANSWER' priority 1 to default

[Mar 10 12:01:12]     -- Added extension 'operator-NOANSWER' priority 1 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension 'operator-BUSY' priority 1 to default

[Mar 10 12:01:12]     -- Added extension 'operator-BUSY' priority 1 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'operator' priority 102 to default

[Mar 10 12:01:12]     -- Added extension 'operator' priority 102 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '0' priority 1 to default

[Mar 10 12:01:12]     -- Added extension '0' priority 1 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '0' priority 2 to default

[Mar 10 12:01:12]     -- Added extension '0' priority 2 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '0' priority 3 to default

[Mar 10 12:01:12]     -- Added extension '0' priority 3 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '1' priority 1 to default

[Mar 10 12:01:12]     -- Added extension '1' priority 1 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '1' priority 2 to default

[Mar 10 12:01:12]     -- Added extension '1' priority 2 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '1' priority 3 to default

[Mar 10 12:01:12]     -- Added extension '1' priority 3 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '1' priority 4 to default

[Mar 10 12:01:12]     -- Added extension '1' priority 4 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '2' priority 1 to default

[Mar 10 12:01:12]     -- Added extension '2' priority 1 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '2' priority 2 to default

[Mar 10 12:01:12]     -- Added extension '2' priority 2 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '2' priority 3 to default

[Mar 10 12:01:12]     -- Added extension '2' priority 3 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '2' priority 4 to default

[Mar 10 12:01:12]     -- Added extension '2' priority 4 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '3' priority 1 to default

[Mar 10 12:01:12]     -- Added extension '3' priority 1 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '3' priority 2 to default

[Mar 10 12:01:12]     -- Added extension '3' priority 2 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '3' priority 3 to default

[Mar 10 12:01:12]     -- Added extension '3' priority 3 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '4' priority 1 to default

[Mar 10 12:01:12]     -- Added extension '4' priority 1 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '4' priority 2 to default

[Mar 10 12:01:12]     -- Added extension '4' priority 2 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '4' priority 3 to default

[Mar 10 12:01:12]     -- Added extension '4' priority 3 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '4' priority 4 to default

[Mar 10 12:01:12]     -- Added extension '4' priority 4 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '22' priority 1 to default

[Mar 10 12:01:12]     -- Added extension '22' priority 1 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '22' priority 2 to default

[Mar 10 12:01:12]     -- Added extension '22' priority 2 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '711' priority 1 to default

[Mar 10 12:01:12]     -- Added extension '711' priority 1 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '711' priority 2 to default

[Mar 10 12:01:12]     -- Added extension '711' priority 2 to default

[Mar 10 12:01:12]     -- Including context 'smvoice-intercom' in context 'default'
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_XXX' priority 1 to default

[Mar 10 12:01:12]     -- Added extension '_XXX' priority 1 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_XXX' priority 2 to default

[Mar 10 12:01:12]     -- Added extension '_XXX' priority 2 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_XXX' priority 3 to default

[Mar 10 12:01:12]     -- Added extension '_XXX' priority 3 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_XXX' priority 4 to default

[Mar 10 12:01:12]     -- Added extension '_XXX' priority 4 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '_*XXX' priority 1 to default

[Mar 10 12:01:12]     -- Added extension '_*XXX' priority 1 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_*XXX' priority 2 to default

[Mar 10 12:01:12]     -- Added extension '_*XXX' priority 2 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_*XXX' priority 3 to default

[Mar 10 12:01:12]     -- Added extension '_*XXX' priority 3 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '7000' priority 1 to default

[Mar 10 12:01:12]     -- Added extension '7000' priority 1 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '7001' priority 1 to default

[Mar 10 12:01:12]     -- Added extension '7001' priority 1 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '7002' priority 1 to default

[Mar 10 12:01:12]     -- Added extension '7002' priority 1 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_XXXXX' priority 1 to default

[Mar 10 12:01:12]     -- Added extension '_XXXXX' priority 1 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_XXXXX' priority 2 to default

[Mar 10 12:01:12]     -- Added extension '_XXXXX' priority 2 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension 'i' priority 1 to default

[Mar 10 12:01:12]     -- Added extension 'i' priority 1 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'i' priority 2 to default

[Mar 10 12:01:12]     -- Added extension 'i' priority 2 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'i' priority 3 to default

[Mar 10 12:01:12]     -- Added extension 'i' priority 3 to default
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:3674 ast_context_create: Registered context 'smvoice-local'

[Mar 10 12:01:12]     -- Registered extension context 'smvoice-local'
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '297' priority 1 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '297' priority 1 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '298' priority 1 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '298' priority 1 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '299' priority 1 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '299' priority 1 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_1XX' priority 1 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_1XX' priority 1 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_1XX' priority 2 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_1XX' priority 2 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_1XX' priority 3 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_1XX' priority 3 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_1XX' priority 4 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_1XX' priority 4 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_1XX' priority 5 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_1XX' priority 5 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_1XX' priority 6 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_1XX' priority 6 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_1XX' priority 7 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_1XX' priority 7 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_1XX-CHANUNAVAIL' priority 1 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_1XX-CHANUNAVAIL' priority 1 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_1XX-CONGESTION' priority 1 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_1XX-CONGESTION' priority 1 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_1XX-NOANSWER' priority 1 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_1XX-NOANSWER' priority 1 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '_1XX-BUSY' priority 1 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_1XX-BUSY' priority 1 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_206' priority 1 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_206' priority 1 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_2XX' priority 1 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_2XX' priority 1 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_2XX' priority 2 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_2XX' priority 2 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_2XX' priority 3 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_2XX' priority 3 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_2XX' priority 4 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_2XX' priority 4 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_2XX' priority 5 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_2XX' priority 5 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_2XX' priority 6 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_2XX' priority 6 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_2XX' priority 7 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_2XX' priority 7 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_2XX-CHANUNAVAIL' priority 1 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_2XX-CHANUNAVAIL' priority 1 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_2XX-CONGESTION' priority 1 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_2XX-CONGESTION' priority 1 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_2XX-NOANSWER' priority 1 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_2XX-NOANSWER' priority 1 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '_2XX-BUSY' priority 1 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_2XX-BUSY' priority 1 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_4XX' priority 1 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_4XX' priority 1 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_4XX' priority 2 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_4XX' priority 2 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_4XX' priority 3 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_4XX' priority 3 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_4XX' priority 4 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_4XX' priority 4 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_4XX-CHANUNAVAIL' priority 1 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_4XX-CHANUNAVAIL' priority 1 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_4XX-CONGESTION' priority 1 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_4XX-CONGESTION' priority 1 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_4XX-NOANSWER' priority 1 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_4XX-NOANSWER' priority 1 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '_4XX-BUSY' priority 1 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_4XX-BUSY' priority 1 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_5XX' priority 1 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_5XX' priority 1 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_5XX' priority 2 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_5XX' priority 2 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_5XX' priority 3 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_5XX' priority 3 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_5XX' priority 4 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_5XX' priority 4 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_5XX' priority 5 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_5XX' priority 5 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_5XX' priority 6 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_5XX' priority 6 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_5XX' priority 7 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_5XX' priority 7 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_5XX-CHANUNAVAIL' priority 1 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_5XX-CHANUNAVAIL' priority 1 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_5XX-CONGESTION' priority 1 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_5XX-CONGESTION' priority 1 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_5XX-NOANSWER' priority 1 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_5XX-NOANSWER' priority 1 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '_5XX-BUSY' priority 1 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_5XX-BUSY' priority 1 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_6XX' priority 1 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_6XX' priority 1 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_6XX' priority 2 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_6XX' priority 2 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_6XX' priority 3 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_6XX' priority 3 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_6XX' priority 4 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_6XX' priority 4 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_6XX-CHANUNAVAIL' priority 1 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_6XX-CHANUNAVAIL' priority 1 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_6XX-CONGESTION' priority 1 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_6XX-CONGESTION' priority 1 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_6XX-NOANSWER' priority 1 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_6XX-NOANSWER' priority 1 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '_6XX-BUSY' priority 1 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_6XX-BUSY' priority 1 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_8XX' priority 1 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_8XX' priority 1 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_8XX' priority 2 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_8XX' priority 2 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_8XX' priority 3 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_8XX' priority 3 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_8XX' priority 4 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_8XX' priority 4 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_8XX-CHANUNAVAIL' priority 1 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_8XX-CHANUNAVAIL' priority 1 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_8XX-CONGESTION' priority 1 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_8XX-CONGESTION' priority 1 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_8XX-NOANSWER' priority 1 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_8XX-NOANSWER' priority 1 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '_8XX-BUSY' priority 1 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_8XX-BUSY' priority 1 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '_*XXX' priority 1 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_*XXX' priority 1 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_*XXX' priority 2 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_*XXX' priority 2 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_*XXX' priority 3 to smvoice-local

[Mar 10 12:01:12]     -- Added extension '_*XXX' priority 3 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension 'INVALID' priority 1 to smvoice-local

[Mar 10 12:01:12]     -- Added extension 'INVALID' priority 1 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'INVALID' priority 2 to smvoice-local

[Mar 10 12:01:12]     -- Added extension 'INVALID' priority 2 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'INVALID' priority 3 to smvoice-local

[Mar 10 12:01:12]     -- Added extension 'INVALID' priority 3 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension 'i' priority 1 to smvoice-local

[Mar 10 12:01:12]     -- Added extension 'i' priority 1 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'i' priority 2 to smvoice-local

[Mar 10 12:01:12]     -- Added extension 'i' priority 2 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'i' priority 3 to smvoice-local

[Mar 10 12:01:12]     -- Added extension 'i' priority 3 to smvoice-local
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:3674 ast_context_create: Registered context 'smvoice-sip'

[Mar 10 12:01:12]     -- Registered extension context 'smvoice-sip'

[Mar 10 12:01:12]     -- Including context 'smvoice-iaxy' in context 'smvoice-sip'
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '82473' priority 1 to smvoice-sip

[Mar 10 12:01:12]     -- Added extension '82473' priority 1 to smvoice-sip
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:3674 ast_context_create: Registered context 'wellgate'

[Mar 10 12:01:12]     -- Registered extension context 'wellgate'
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension 's' priority 1 to wellgate

[Mar 10 12:01:12]     -- Added extension 's' priority 1 to wellgate
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '801' priority 1 to wellgate

[Mar 10 12:01:12]     -- Added extension '801' priority 1 to wellgate
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '801' priority 2 to wellgate

[Mar 10 12:01:12]     -- Added extension '801' priority 2 to wellgate
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '802' priority 1 to wellgate

[Mar 10 12:01:12]     -- Added extension '802' priority 1 to wellgate
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '802' priority 2 to wellgate

[Mar 10 12:01:12]     -- Added extension '802' priority 2 to wellgate
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '803' priority 1 to wellgate

[Mar 10 12:01:12]     -- Added extension '803' priority 1 to wellgate
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '803' priority 2 to wellgate

[Mar 10 12:01:12]     -- Added extension '803' priority 2 to wellgate
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '804' priority 1 to wellgate

[Mar 10 12:01:12]     -- Added extension '804' priority 1 to wellgate
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '804' priority 2 to wellgate

[Mar 10 12:01:12]     -- Added extension '804' priority 2 to wellgate
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:3674 ast_context_create: Registered context 'smvoice-iaxy'

[Mar 10 12:01:12]     -- Registered extension context 'smvoice-iaxy'
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '50' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '50' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '50' priority 2 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '50' priority 2 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '55' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '55' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '56' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '56' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '57' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '57' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '57' priority 2 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '57' priority 2 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '57' priority 3 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '57' priority 3 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '57' priority 4 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '57' priority 4 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '58' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '58' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '58' priority 2 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '58' priority 2 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '58' priority 3 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '58' priority 3 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '58' priority 4 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '58' priority 4 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '59' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '59' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '5068012' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '5068012' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '199' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '199' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '1041' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '1041' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '1041' priority 2 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '1041' priority 2 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '1104' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '1104' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '1104' priority 2 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '1104' priority 2 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '10000' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '10000' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] WARNING[15131]: pbx.c:4796 ast_add_extension2: Unable to register extension '59', priority 1 in 'smvoice-iaxy', already in use
[Mar 10 12:01:12] WARNING[15131]: pbx_config.c:1753 pbx_load_module: Unable to register extension at line 539
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '59' priority 2 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '59' priority 2 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '59' priority 3 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '59' priority 3 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '59' priority 4 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '59' priority 4 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '*70' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '*70' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '*71' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '*71' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '*72' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '*72' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '*73' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '*73' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '*74' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '*74' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '*75' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '*75' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '*76' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '*76' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '*77' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '*77' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '86' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '86' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '777' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '777' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '777' priority 2 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '777' priority 2 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '777' priority 3 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '777' priority 3 to smvoice-iaxy

[Mar 10 12:01:12]     -- Including context 'smvoice-intercom' in context 'smvoice-iaxy'

[Mar 10 12:01:12]     -- Including context 'smvoice-local' in context 'smvoice-iaxy'
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '7000' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '7000' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '7001' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '7001' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '7002' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '7002' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_9XXXXXXX' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_9XXXXXXX' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_9XXXXXXX' priority 2 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_9XXXXXXX' priority 2 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_9XXXXXXX-CHANUNAVAIL' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_9XXXXXXX-CHANUNAVAIL' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_9XXXXXXX-CHANUNAVAIL' priority 2 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_9XXXXXXX-CHANUNAVAIL' priority 2 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_9XXXXXXX-CONGESTION' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_9XXXXXXX-CONGESTION' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '_9XXXXXXX-BUSY' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_9XXXXXXX-BUSY' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_9XXXXXXXXXX' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_9XXXXXXXXXX' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_9XXXXXXXXXX' priority 2 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_9XXXXXXXXXX' priority 2 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_9XXXXXXXXXX-CHANUNAVAIL' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_9XXXXXXXXXX-CHANUNAVAIL' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_9XXXXXXXXXX-CHANUNAVAIL' priority 2 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_9XXXXXXXXXX-CHANUNAVAIL' priority 2 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '_9XXXXXXXXXX-CONGESTION' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_9XXXXXXXXXX-CONGESTION' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '_9XXXXXXXXXX-BUSY' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_9XXXXXXXXXX-BUSY' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '_91XXXXXXXXXX' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_91XXXXXXXXXX' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_91XXXXXXXXXX' priority 2 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_91XXXXXXXXXX' priority 2 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '_91XXXXXXXXXX-CHANUNAVAIL' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_91XXXXXXXXXX-CHANUNAVAIL' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_91XXXXXXXXXX-CHANUNAVAIL' priority 2 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_91XXXXXXXXXX-CHANUNAVAIL' priority 2 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '_91XXXXXXXXXX-CONGESTION' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_91XXXXXXXXXX-CONGESTION' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '_91XXXXXXXXXX-BUSY' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_91XXXXXXXXXX-BUSY' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '_8XXXXXXX' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_8XXXXXXX' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_8XXXXXXX' priority 2 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_8XXXXXXX' priority 2 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_8XXXXXXX' priority 3 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_8XXXXXXX' priority 3 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '_8XXXXXXX-CHANUNAVAIL' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_8XXXXXXX-CHANUNAVAIL' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_8XXXXXXX-CHANUNAVAIL' priority 2 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_8XXXXXXX-CHANUNAVAIL' priority 2 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '_8XXXXXXX-CONGESTION' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_8XXXXXXX-CONGESTION' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_8XXXXXXX-CONGESTION' priority 2 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_8XXXXXXX-CONGESTION' priority 2 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '_8XXXXXXXXXX' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_8XXXXXXXXXX' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_8XXXXXXXXXX' priority 2 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_8XXXXXXXXXX' priority 2 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_8XXXXXXXXXX' priority 3 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_8XXXXXXXXXX' priority 3 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '_8XXXXXXXXXX-CHANUNAVAIL' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_8XXXXXXXXXX-CHANUNAVAIL' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_8XXXXXXXXXX-CHANUNAVAIL' priority 2 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_8XXXXXXXXXX-CHANUNAVAIL' priority 2 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '_8XXXXXXXXXX-CONGESTION' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_8XXXXXXXXXX-CONGESTION' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_8XXXXXXXXXX-CONGESTION' priority 2 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_8XXXXXXXXXX-CONGESTION' priority 2 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '_81XXXXXXXXXX' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_81XXXXXXXXXX' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_81XXXXXXXXXX' priority 2 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_81XXXXXXXXXX' priority 2 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_81XXXXXXXXXX' priority 3 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_81XXXXXXXXXX' priority 3 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '_81XXXXXXXXXX-CHANUNAVAIL' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_81XXXXXXXXXX-CHANUNAVAIL' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_81XXXXXXXXXX-CHANUNAVAIL' priority 2 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_81XXXXXXXXXX-CHANUNAVAIL' priority 2 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '_81XXXXXXXXXX-CONGESTION' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_81XXXXXXXXXX-CONGESTION' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_81XXXXXXXXXX-CONGESTION' priority 2 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_81XXXXXXXXXX-CONGESTION' priority 2 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '_7XXXXXXX' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_7XXXXXXX' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_7XXXXXXX' priority 2 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_7XXXXXXX' priority 2 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_7XXXXXXX' priority 3 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_7XXXXXXX' priority 3 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '_71XXXXXXXXXX' priority 1 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_71XXXXXXXXXX' priority 1 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '_7XXXXXXXXXXX' priority 2 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_7XXXXXXXXXXX' priority 2 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '_71XXXXXXXXXX' priority 3 to smvoice-iaxy

[Mar 10 12:01:12]     -- Added extension '_71XXXXXXXXXX' priority 3 to smvoice-iaxy
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:3674 ast_context_create: Registered context 'smvoice-return-voicemail'

[Mar 10 12:01:12]     -- Registered extension context 'smvoice-return-voicemail'
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension 's' priority 1 to smvoice-return-voicemail

[Mar 10 12:01:12]     -- Added extension 's' priority 1 to smvoice-return-voicemail
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:3674 ast_context_create: Registered context 'smvoice-voicemail'

[Mar 10 12:01:12]     -- Registered extension context 'smvoice-voicemail'
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '0' priority 1 to smvoice-voicemail

[Mar 10 12:01:12]     -- Added extension '0' priority 1 to smvoice-voicemail
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '1' priority 1 to smvoice-voicemail

[Mar 10 12:01:12]     -- Added extension '1' priority 1 to smvoice-voicemail
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '2' priority 1 to smvoice-voicemail

[Mar 10 12:01:12]     -- Added extension '2' priority 1 to smvoice-voicemail
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '3' priority 1 to smvoice-voicemail

[Mar 10 12:01:12]     -- Added extension '3' priority 1 to smvoice-voicemail
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '101' priority 1 to smvoice-voicemail

[Mar 10 12:01:12]     -- Added extension '101' priority 1 to smvoice-voicemail
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '510' priority 1 to smvoice-voicemail

[Mar 10 12:01:12]     -- Added extension '510' priority 1 to smvoice-voicemail
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '511' priority 1 to smvoice-voicemail

[Mar 10 12:01:12]     -- Added extension '511' priority 1 to smvoice-voicemail
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '512' priority 1 to smvoice-voicemail

[Mar 10 12:01:12]     -- Added extension '512' priority 1 to smvoice-voicemail
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '513' priority 1 to smvoice-voicemail

[Mar 10 12:01:12]     -- Added extension '513' priority 1 to smvoice-voicemail
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '514' priority 1 to smvoice-voicemail

[Mar 10 12:01:12]     -- Added extension '514' priority 1 to smvoice-voicemail
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '515' priority 1 to smvoice-voicemail

[Mar 10 12:01:12]     -- Added extension '515' priority 1 to smvoice-voicemail
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '806' priority 1 to smvoice-voicemail

[Mar 10 12:01:12]     -- Added extension '806' priority 1 to smvoice-voicemail
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '205' priority 1 to smvoice-voicemail

[Mar 10 12:01:12]     -- Added extension '205' priority 1 to smvoice-voicemail
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '522' priority 1 to smvoice-voicemail

[Mar 10 12:01:12]     -- Added extension '522' priority 1 to smvoice-voicemail
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '605' priority 1 to smvoice-voicemail

[Mar 10 12:01:12]     -- Added extension '605' priority 1 to smvoice-voicemail
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '801' priority 1 to smvoice-voicemail

[Mar 10 12:01:12]     -- Added extension '801' priority 1 to smvoice-voicemail
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '210' priority 1 to smvoice-voicemail

[Mar 10 12:01:12]     -- Added extension '210' priority 1 to smvoice-voicemail
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '216' priority 1 to smvoice-voicemail

[Mar 10 12:01:12]     -- Added extension '216' priority 1 to smvoice-voicemail
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '592' priority 1 to smvoice-voicemail

[Mar 10 12:01:12]     -- Added extension '592' priority 1 to smvoice-voicemail
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '2134' priority 1 to smvoice-voicemail

[Mar 10 12:01:12]     -- Added extension '2134' priority 1 to smvoice-voicemail
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '209' priority 1 to smvoice-voicemail

[Mar 10 12:01:12]     -- Added extension '209' priority 1 to smvoice-voicemail
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '204' priority 1 to smvoice-voicemail

[Mar 10 12:01:12]     -- Added extension '204' priority 1 to smvoice-voicemail
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '401' priority 1 to smvoice-voicemail

[Mar 10 12:01:12]     -- Added extension '401' priority 1 to smvoice-voicemail
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '402' priority 1 to smvoice-voicemail

[Mar 10 12:01:12]     -- Added extension '402' priority 1 to smvoice-voicemail
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '403' priority 1 to smvoice-voicemail

[Mar 10 12:01:12]     -- Added extension '403' priority 1 to smvoice-voicemail
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '404' priority 1 to smvoice-voicemail

[Mar 10 12:01:12]     -- Added extension '404' priority 1 to smvoice-voicemail
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '528' priority 1 to smvoice-voicemail

[Mar 10 12:01:12]     -- Added extension '528' priority 1 to smvoice-voicemail
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '530' priority 1 to smvoice-voicemail

[Mar 10 12:01:12]     -- Added extension '530' priority 1 to smvoice-voicemail
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '531' priority 1 to smvoice-voicemail

[Mar 10 12:01:12]     -- Added extension '531' priority 1 to smvoice-voicemail
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '606' priority 1 to smvoice-voicemail

[Mar 10 12:01:12]     -- Added extension '606' priority 1 to smvoice-voicemail
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '616' priority 1 to smvoice-voicemail

[Mar 10 12:01:12]     -- Added extension '616' priority 1 to smvoice-voicemail
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension '800' priority 1 to smvoice-voicemail

[Mar 10 12:01:12]     -- Added extension '800' priority 1 to smvoice-voicemail
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension 'i' priority 1 to smvoice-voicemail

[Mar 10 12:01:12]     -- Added extension 'i' priority 1 to smvoice-voicemail
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'i' priority 2 to smvoice-voicemail

[Mar 10 12:01:12]     -- Added extension 'i' priority 2 to smvoice-voicemail
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:3674 ast_context_create: Registered context 'smvoice-intercom'

[Mar 10 12:01:12]     -- Registered extension context 'smvoice-intercom'
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension '87' priority 1 to smvoice-intercom

[Mar 10 12:01:12]     -- Added extension '87' priority 1 to smvoice-intercom
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension '87' priority 2 to smvoice-intercom

[Mar 10 12:01:12]     -- Added extension '87' priority 2 to smvoice-intercom
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:3674 ast_context_create: Registered context 'smvoice-incoming'

[Mar 10 12:01:12]     -- Registered extension context 'smvoice-incoming'
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension 's' priority 1 to smvoice-incoming

[Mar 10 12:01:12]     -- Added extension 's' priority 1 to smvoice-incoming

[Mar 10 12:01:12]     -- Including context 'smvoice-intercom' in context 'smvoice-incoming'

[Mar 10 12:01:12]     -- Including context 'smvoice-transfers' in context 'smvoice-incoming'
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:3674 ast_context_create: Registered context 'smvoice-incoming-kx1232'

[Mar 10 12:01:12]     -- Registered extension context 'smvoice-incoming-kx1232'
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension 's' priority 1 to smvoice-incoming-kx1232

[Mar 10 12:01:12]     -- Added extension 's' priority 1 to smvoice-incoming-kx1232
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 's' priority 2 to smvoice-incoming-kx1232

[Mar 10 12:01:12]     -- Added extension 's' priority 2 to smvoice-incoming-kx1232
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 's' priority 3 to smvoice-incoming-kx1232

[Mar 10 12:01:12]     -- Added extension 's' priority 3 to smvoice-incoming-kx1232
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:3674 ast_context_create: Registered context 'smvoice-transfers'

[Mar 10 12:01:12]     -- Registered extension context 'smvoice-transfers'
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension 'smvoice_dial' priority 1 to smvoice-transfers

[Mar 10 12:01:12]     -- Added extension 'smvoice_dial' priority 1 to smvoice-transfers
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'smvoice_dial' priority 2 to smvoice-transfers

[Mar 10 12:01:12]     -- Added extension 'smvoice_dial' priority 2 to smvoice-transfers
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension 'smvoice_dial-CHANUNAVAIL' priority 1 to smvoice-transfers

[Mar 10 12:01:12]     -- Added extension 'smvoice_dial-CHANUNAVAIL' priority 1 to smvoice-transfers
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'smvoice_dial-CHANUNAVAIL' priority 2 to smvoice-transfers

[Mar 10 12:01:12]     -- Added extension 'smvoice_dial-CHANUNAVAIL' priority 2 to smvoice-transfers
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension 'smvoice_dial-CONGESTION' priority 1 to smvoice-transfers

[Mar 10 12:01:12]     -- Added extension 'smvoice_dial-CONGESTION' priority 1 to smvoice-transfers
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'smvoice_dial-CONGESTION' priority 2 to smvoice-transfers

[Mar 10 12:01:12]     -- Added extension 'smvoice_dial-CONGESTION' priority 2 to smvoice-transfers
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension 'smvoice_dial-NOANSWER' priority 1 to smvoice-transfers

[Mar 10 12:01:12]     -- Added extension 'smvoice_dial-NOANSWER' priority 1 to smvoice-transfers
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'smvoice_dial-NOANSWER' priority 2 to smvoice-transfers

[Mar 10 12:01:12]     -- Added extension 'smvoice_dial-NOANSWER' priority 2 to smvoice-transfers
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension 'smvoice_dial-BUSY' priority 1 to smvoice-transfers

[Mar 10 12:01:12]     -- Added extension 'smvoice_dial-BUSY' priority 1 to smvoice-transfers
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'smvoice_dial-BUSY' priority 2 to smvoice-transfers

[Mar 10 12:01:12]     -- Added extension 'smvoice_dial-BUSY' priority 2 to smvoice-transfers
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension 'smvoice_dial_no_extension' priority 1 to smvoice-transfers

[Mar 10 12:01:12]     -- Added extension 'smvoice_dial_no_extension' priority 1 to smvoice-transfers
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'smvoice_dial_no_extension' priority 2 to smvoice-transfers

[Mar 10 12:01:12]     -- Added extension 'smvoice_dial_no_extension' priority 2 to smvoice-transfers
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension 'smvoice_dial_no_extension-CHANUNAVAIL' priority 1 to smvoice-transfers

[Mar 10 12:01:12]     -- Added extension 'smvoice_dial_no_extension-CHANUNAVAIL' priority 1 to smvoice-transfers
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'smvoice_dial_no_extension-CHANUNAVAIL' priority 2 to smvoice-transfers

[Mar 10 12:01:12]     -- Added extension 'smvoice_dial_no_extension-CHANUNAVAIL' priority 2 to smvoice-transfers
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension 'smvoice_dial_no_extension-CONGESTION' priority 1 to smvoice-transfers

[Mar 10 12:01:12]     -- Added extension 'smvoice_dial_no_extension-CONGESTION' priority 1 to smvoice-transfers
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension 'smvoice_dial_no_extension-BUSY' priority 1 to smvoice-transfers

[Mar 10 12:01:12]     -- Added extension 'smvoice_dial_no_extension-BUSY' priority 1 to smvoice-transfers
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension 'smvoice_dial_no_extension-NOANSWER' priority 1 to smvoice-transfers

[Mar 10 12:01:12]     -- Added extension 'smvoice_dial_no_extension-NOANSWER' priority 1 to smvoice-transfers
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'smvoice_dial_no_extension-NOANSWER' priority 2 to smvoice-transfers

[Mar 10 12:01:12]     -- Added extension 'smvoice_dial_no_extension-NOANSWER' priority 2 to smvoice-transfers
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension 'smvoice_dial_goto_voicemail' priority 1 to smvoice-transfers

[Mar 10 12:01:12]     -- Added extension 'smvoice_dial_goto_voicemail' priority 1 to smvoice-transfers
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'smvoice_dial_goto_voicemail' priority 2 to smvoice-transfers

[Mar 10 12:01:12]     -- Added extension 'smvoice_dial_goto_voicemail' priority 2 to smvoice-transfers
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension 'smvoice_dial_goto_voicemail-CHANUNAVAIL' priority 1 to smvoice-transfers

[Mar 10 12:01:12]     -- Added extension 'smvoice_dial_goto_voicemail-CHANUNAVAIL' priority 1 to smvoice-transfers
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension 'smvoice_dial_goto_voicemail-CONGESTION' priority 1 to smvoice-transfers

[Mar 10 12:01:12]     -- Added extension 'smvoice_dial_goto_voicemail-CONGESTION' priority 1 to smvoice-transfers
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension 'smvoice_dial_goto_voicemail-NOANSWER' priority 1 to smvoice-transfers

[Mar 10 12:01:12]     -- Added extension 'smvoice_dial_goto_voicemail-NOANSWER' priority 1 to smvoice-transfers
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension 'smvoice_dial_goto_voicemail-BUSY' priority 1 to smvoice-transfers

[Mar 10 12:01:12]     -- Added extension 'smvoice_dial_goto_voicemail-BUSY' priority 1 to smvoice-transfers
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension 'smvoice_conference' priority 1 to smvoice-transfers

[Mar 10 12:01:12]     -- Added extension 'smvoice_conference' priority 1 to smvoice-transfers
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension 'smvoice_sendtext' priority 1 to smvoice-transfers

[Mar 10 12:01:12]     -- Added extension 'smvoice_sendtext' priority 1 to smvoice-transfers
[Mar 10 12:01:12] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'smvoice_sendtext' priority 2 to smvoice-transfers

[Mar 10 12:01:14]     -- Added extension 'smvoice_sendtext' priority 2 to smvoice-transfers
[Mar 10 12:01:14] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'smvoice_sendtext' priority 3 to smvoice-transfers

[Mar 10 12:01:14]     -- Added extension 'smvoice_sendtext' priority 3 to smvoice-transfers
[Mar 10 12:01:14] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension 'smvoice_no_callprogress' priority 1 to smvoice-transfers

[Mar 10 12:01:14]     -- Added extension 'smvoice_no_callprogress' priority 1 to smvoice-transfers
[Mar 10 12:01:14] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension 'smvoice_callprogress' priority 1 to smvoice-transfers

[Mar 10 12:01:14]     -- Added extension 'smvoice_callprogress' priority 1 to smvoice-transfers
[Mar 10 12:01:14] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'smvoice_callprogress' priority 2 to smvoice-transfers

[Mar 10 12:01:14]     -- Added extension 'smvoice_callprogress' priority 2 to smvoice-transfers
[Mar 10 12:01:14] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'smvoice_callprogress' priority 3 to smvoice-transfers

[Mar 10 12:01:14]     -- Added extension 'smvoice_callprogress' priority 3 to smvoice-transfers
[Mar 10 12:01:14] DEBUG[15131]: pbx.c:3674 ast_context_create: Registered context 'smvoice-homework-hotline'

[Mar 10 12:01:14]     -- Registered extension context 'smvoice-homework-hotline'
[Mar 10 12:01:14] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension 's' priority 1 to smvoice-homework-hotline

[Mar 10 12:01:14]     -- Added extension 's' priority 1 to smvoice-homework-hotline
[Mar 10 12:01:14] DEBUG[15131]: pbx.c:3674 ast_context_create: Registered context 'smvoice-faxout'

[Mar 10 12:01:14]     -- Registered extension context 'smvoice-faxout'
[Mar 10 12:01:14] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension 'smvoice_faxout' priority 1 to smvoice-faxout

[Mar 10 12:01:14]     -- Added extension 'smvoice_faxout' priority 1 to smvoice-faxout
[Mar 10 12:01:14] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'smvoice_faxout' priority 2 to smvoice-faxout

[Mar 10 12:01:14]     -- Added extension 'smvoice_faxout' priority 2 to smvoice-faxout
[Mar 10 12:01:14] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'smvoice_faxout' priority 3 to smvoice-faxout

[Mar 10 12:01:14]     -- Added extension 'smvoice_faxout' priority 3 to smvoice-faxout
[Mar 10 12:01:14] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'smvoice_faxout' priority 4 to smvoice-faxout

[Mar 10 12:01:14]     -- Added extension 'smvoice_faxout' priority 4 to smvoice-faxout
[Mar 10 12:01:14] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'smvoice_faxout' priority 104 to smvoice-faxout

[Mar 10 12:01:14]     -- Added extension 'smvoice_faxout' priority 104 to smvoice-faxout
[Mar 10 12:01:14] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'smvoice_faxout' priority 105 to smvoice-faxout

[Mar 10 12:01:14]     -- Added extension 'smvoice_faxout' priority 105 to smvoice-faxout
[Mar 10 12:01:14] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension 'i' priority 1 to smvoice-faxout

[Mar 10 12:01:14]     -- Added extension 'i' priority 1 to smvoice-faxout
[Mar 10 12:01:14] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'i' priority 2 to smvoice-faxout

[Mar 10 12:01:14]     -- Added extension 'i' priority 2 to smvoice-faxout
[Mar 10 12:01:14] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension 't' priority 1 to smvoice-faxout

[Mar 10 12:01:14]     -- Added extension 't' priority 1 to smvoice-faxout
[Mar 10 12:01:14] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 't' priority 2 to smvoice-faxout

[Mar 10 12:01:14]     -- Added extension 't' priority 2 to smvoice-faxout
[Mar 10 12:01:14] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension 'T' priority 1 to smvoice-faxout

[Mar 10 12:01:14]     -- Added extension 'T' priority 1 to smvoice-faxout
[Mar 10 12:01:14] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'T' priority 2 to smvoice-faxout

[Mar 10 12:01:14]     -- Added extension 'T' priority 2 to smvoice-faxout
[Mar 10 12:01:14] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension 'failed' priority 1 to smvoice-faxout

[Mar 10 12:01:14]     -- Added extension 'failed' priority 1 to smvoice-faxout
[Mar 10 12:01:14] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'failed' priority 2 to smvoice-faxout

[Mar 10 12:01:14]     -- Added extension 'failed' priority 2 to smvoice-faxout
[Mar 10 12:01:14] DEBUG[15131]: pbx.c:3674 ast_context_create: Registered context 'smvoice-dialout'

[Mar 10 12:01:14]     -- Registered extension context 'smvoice-dialout'

[Mar 10 12:01:14]     -- Including context 'smvoice-intercom' in context 'smvoice-dialout'

[Mar 10 12:01:14]     -- Including context 'smvoice-transfers' in context 'smvoice-dialout'
[Mar 10 12:01:14] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension 'i' priority 1 to smvoice-dialout

[Mar 10 12:01:14]     -- Added extension 'i' priority 1 to smvoice-dialout
[Mar 10 12:01:14] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension 't' priority 1 to smvoice-dialout

[Mar 10 12:01:14]     -- Added extension 't' priority 1 to smvoice-dialout
[Mar 10 12:01:14] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension 'T' priority 1 to smvoice-dialout

[Mar 10 12:01:14]     -- Added extension 'T' priority 1 to smvoice-dialout
[Mar 10 12:01:14] DEBUG[15131]: pbx.c:4861 ast_add_extension2: Added extension 'failed' priority 1 to smvoice-dialout

[Mar 10 12:01:14]     -- Added extension 'failed' priority 1 to smvoice-dialout
[Mar 10 12:01:14] DEBUG[15131]: pbx.c:3674 ast_context_create: Registered context 'iax_devcentos64_to_unifiedpaging'

[Mar 10 12:01:14]     -- Registered extension context 'iax_devcentos64_to_unifiedpaging'
[Mar 10 12:01:14] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension 's' priority 1 to iax_devcentos64_to_unifiedpaging

[Mar 10 12:01:14]     -- Added extension 's' priority 1 to iax_devcentos64_to_unifiedpaging
[Mar 10 12:01:14] DEBUG[15131]: pbx.c:3674 ast_context_create: Registered context 'smvoice-testing'

[Mar 10 12:01:14]     -- Registered extension context 'smvoice-testing'
[Mar 10 12:01:14] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension 'call_cell' priority 1 to smvoice-testing

[Mar 10 12:01:14]     -- Added extension 'call_cell' priority 1 to smvoice-testing
[Mar 10 12:01:14] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'call_cell' priority 2 to smvoice-testing

[Mar 10 12:01:14]     -- Added extension 'call_cell' priority 2 to smvoice-testing
[Mar 10 12:01:14] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'call_cell' priority 3 to smvoice-testing

[Mar 10 12:01:14]     -- Added extension 'call_cell' priority 3 to smvoice-testing
[Mar 10 12:01:14] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'call_cell' priority 4 to smvoice-testing

[Mar 10 12:01:14]     -- Added extension 'call_cell' priority 4 to smvoice-testing
[Mar 10 12:01:14] DEBUG[15131]: pbx.c:4876 ast_add_extension2: Added extension 'failed' priority 1 to smvoice-testing

[Mar 10 12:01:14]     -- Added extension 'failed' priority 1 to smvoice-testing
[Mar 10 12:01:14] DEBUG[15131]: pbx.c:4842 ast_add_extension2: Added extension 'failed' priority 2 to smvoice-testing

[Mar 10 12:01:14]     -- Added extension 'failed' priority 2 to smvoice-testing

[Mar 10 12:01:14]  [pbx_functions.so][Mar 10 12:01:14]  => (Builtin dialplan functions)

[Mar 10 12:01:14]   == Registered custom function MD5

[Mar 10 12:01:14]   == Registered custom function CHECK_MD5

[Mar 10 12:01:14]   == Registered custom function MATH

[Mar 10 12:01:14]   == Registered custom function GROUP_COUNT

[Mar 10 12:01:14]   == Registered custom function GROUP_MATCH_COUNT

[Mar 10 12:01:14]   == Registered custom function GROUP

[Mar 10 12:01:14]   == Registered custom function GROUP_LIST

[Mar 10 12:01:14]   == Registered custom function FIELDQTY

[Mar 10 12:01:14]   == Registered custom function REGEX

[Mar 10 12:01:14]   == Registered custom function LEN

[Mar 10 12:01:14]   == Registered custom function STRFTIME

[Mar 10 12:01:14]   == Registered custom function EVAL

[Mar 10 12:01:14]   == Registered custom function CDR

[Mar 10 12:01:14]   == Registered custom function ISNULL

[Mar 10 12:01:14]   == Registered custom function SET

[Mar 10 12:01:14]   == Registered custom function EXISTS

[Mar 10 12:01:14]   == Registered custom function IF

[Mar 10 12:01:14]   == Registered custom function IFTIME

[Mar 10 12:01:14]   == Registered custom function ENV

[Mar 10 12:01:14]   == Registered custom function DB

[Mar 10 12:01:14]   == Registered custom function DB_EXISTS

[Mar 10 12:01:14]   == Registered custom function TIMEOUT

[Mar 10 12:01:14]   == Registered custom function LANGUAGE

[Mar 10 12:01:14]   == Registered custom function MUSICCLASS

[Mar 10 12:01:14]  [pbx_realtime.so][Mar 10 12:01:14]  => (Realtime Switch)

[Mar 10 12:01:14]  [skipping chan_alsa.so]

[Mar 10 12:01:14]  [chan_features.so][Mar 10 12:01:14]  => (Feature Proxy Channel)
[Mar 10 12:01:14] DEBUG[15131]: channel.c:344 ast_channel_register: Registered handler for 'Feature' (Feature Proxy Channel Driver)

[Mar 10 12:01:14]   == Registered channel type 'Feature' (Feature Proxy Channel Driver)

[Mar 10 12:01:14]  [chan_skinny.so][Mar 10 12:01:14]  => (Skinny Client Control Protocol (Skinny))

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/skinny.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/skinny.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]   == Skinny listening on 0.0.0.0:2000
[Mar 10 12:01:14] DEBUG[15131]: channel.c:344 ast_channel_register: Registered handler for 'Skinny' (Skinny Client Control Protocol (Skinny))

[Mar 10 12:01:14]   == Registered channel type 'Skinny' (Skinny Client Control Protocol (Skinny))

[Mar 10 12:01:14]  [chan_phone.so][Mar 10 12:01:14]  => (Linux Telephony API Support)

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/phone.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/phone.conf
[Mar 10 12:01:14] Found
[Mar 10 12:01:14] DEBUG[15131]: channel.c:344 ast_channel_register: Registered handler for 'Phone' (Standard Linux Telephony API Driver)

[Mar 10 12:01:14]   == Registered channel type 'Phone' (Standard Linux Telephony API Driver)

[Mar 10 12:01:14]  [chan_mgcp.so][Mar 10 12:01:14]  => (Media Gateway Control Protocol (MGCP))

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/mgcp.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/mgcp.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]   == MGCP Listening on 0.0.0.0:2727

[Mar 10 12:01:14]   == Using TOS bits 0
[Mar 10 12:01:14] DEBUG[15131]: channel.c:344 ast_channel_register: Registered handler for 'MGCP' (Media Gateway Control Protocol (MGCP))

[Mar 10 12:01:14]   == Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP))

[Mar 10 12:01:14]  [chan_iax2.so][Mar 10 12:01:14]  => (Inter Asterisk eXchange (Ver 2))

[Mar 10 12:01:14]   == Registered custom function IAXPEER

[Mar 10 12:01:14]   == Registered application 'IAX2Provision'

[Mar 10 12:01:14]   == Manager registered action IAXpeers

[Mar 10 12:01:14]   == Manager registered action IAXnetstats

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/iax.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/iax.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/express.demonstration.extensions.iax.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.iax.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/express.demonstration.extensions.iax.605.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.iax.605.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/express.demonstration.extensions.iax.606.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.iax.606.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/express.demonstration.extensions.iax.614.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.iax.614.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/express.demonstration.extensions.iax.616.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.iax.616.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/express.demonstration.extensions.iax.651.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.iax.651.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/express.demonstration.extensions.iax.698.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.iax.698.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]   == Using TOS bits 16

[Mar 10 12:01:14]   == Binding IAX2 to '24.123.23.170:4569'

[Mar 10 12:01:14]     -- doing lookup for '192.168.1.10'

[Mar 10 12:01:14]     -- doing lookup for '192.168.1.10'
[Mar 10 12:01:14] DEBUG[15131]: db.c:200 ast_db_get: Unable to find key 'dell_8200_to_unifiedpaging' in family 'IAX/Registry'

[Mar 10 12:01:14]     -- doing lookup for 'switch-1.nufone.net'

[Mar 10 12:01:14]     -- doing lookup for '192.168.1.170'

[Mar 10 12:01:14]     -- doing lookup for '192.168.1.159'
[Mar 10 12:01:14] DEBUG[15131]: db.c:200 ast_db_get: Unable to find key '605' in family 'IAX/Registry'

[Mar 10 12:01:14]     -- Seeding '606' at 192.168.1.166:4569 for 60
[Mar 10 12:01:14] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for IAX2/606 - state 4 (Invalid)

[Mar 10 12:01:14]     -- Seeding '614' at 192.168.1.167:4569 for 60
[Mar 10 12:01:14] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for IAX2/614 - state 4 (Invalid)

[Mar 10 12:01:14]     -- Seeding '616' at 74.133.32.69:4569 for 60
[Mar 10 12:01:14] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for IAX2/616 - state 4 (Invalid)

[Mar 10 12:01:14]     -- Seeding '651' at 192.168.1.169:4569 for 60
[Mar 10 12:01:14] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for IAX2/651 - state 4 (Invalid)
[Mar 10 12:01:14] DEBUG[15131]: db.c:200 ast_db_get: Unable to find key '698' in family 'IAX/Registry'
[Mar 10 12:01:14] DEBUG[15131]: channel.c:344 ast_channel_register: Registered handler for 'IAX2' (Inter Asterisk eXchange Driver (Ver 2))

[Mar 10 12:01:14]   == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2))

[Mar 10 12:01:14]   == IAX Ready and Listening

[Mar 10 12:01:14]   == Loaded firmware 'iaxy.bin'

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/iaxprov.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/iaxprov.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]     -- Loaded provisioning template 'default'

[Mar 10 12:01:14]  [chan_sip.so][Mar 10 12:01:14]  => (Session Initiation Protocol (SIP))

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/sip.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/sip.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/express.demonstration.extensions.sip.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/express.demonstration.extensions.sip.520.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.520.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/express.demonstration.extensions.sip.521.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.521.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/express.demonstration.extensions.sip.522.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.522.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/express.demonstration.extensions.sip.523.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.523.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/express.demonstration.extensions.sip.524.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.524.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/express.demonstration.extensions.sip.525.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.525.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/express.demonstration.extensions.sip.526.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.526.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/express.demonstration.extensions.sip.528.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.528.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/express.demonstration.extensions.sip.529.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.529.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/express.demonstration.extensions.sip.540.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.540.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/express.demonstration.extensions.sip.541.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.541.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/express.demonstration.extensions.sip.550.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.550.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/express.demonstration.extensions.sip.551.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.551.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/express.demonstration.extensions.sip.592.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.592.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/express.demonstration.extensions.sip.593.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.593.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/express.demonstration.extensions.sip.594.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.594.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/express.demonstration.extensions.sip.595.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.595.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/express.demonstration.extensions.sip.596.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.596.conf
[Mar 10 12:01:14] Found
[Mar 10 12:01:14] WARNING[15131]: acl.c:182 ast_append_ha: sip.broadvoice.com is not a valid IP
[Mar 10 12:01:14] WARNING[15131]: acl.c:182 ast_append_ha: sip.broadvoice.com is not a valid IP

[Mar 10 12:01:14]     -- SIP Seeding peer from astdb: '510' at 510@192.168.1.165:5060 for 60

[Mar 10 12:01:14]     -- SIP Seeding peer from astdb: '511' at 511@192.168.1.165:5060 for 60

[Mar 10 12:01:14]     -- SIP Seeding peer from astdb: '512' at 512@192.168.1.165:5060 for 60

[Mar 10 12:01:14]     -- SIP Seeding peer from astdb: '513' at 513@192.168.1.165:5060 for 60

[Mar 10 12:01:14]     -- SIP Seeding peer from astdb: '514' at 514@192.168.1.165:5060 for 60

[Mar 10 12:01:14]     -- SIP Seeding peer from astdb: '515' at 515@192.168.1.165:5060 for 60
[Mar 10 12:01:14] DEBUG[15131]: db.c:200 ast_db_get: Unable to find key '520' in family 'SIP/Registry'
[Mar 10 12:01:14] DEBUG[15131]: db.c:200 ast_db_get: Unable to find key '530' in family 'SIP/Registry'
[Mar 10 12:01:14] DEBUG[15131]: db.c:200 ast_db_get: Unable to find key '531' in family 'SIP/Registry'
[Mar 10 12:01:14] DEBUG[15131]: db.c:200 ast_db_get: Unable to find key '532' in family 'SIP/Registry'
[Mar 10 12:01:14] DEBUG[15131]: db.c:200 ast_db_get: Unable to find key '534' in family 'SIP/Registry'
[Mar 10 12:01:14] DEBUG[15131]: db.c:200 ast_db_get: Unable to find key '535' in family 'SIP/Registry'
[Mar 10 12:01:14] DEBUG[15131]: db.c:200 ast_db_get: Unable to find key '536' in family 'SIP/Registry'
[Mar 10 12:01:14] DEBUG[15131]: db.c:200 ast_db_get: Unable to find key '537' in family 'SIP/Registry'

[Mar 10 12:01:14]     -- SIP Seeding peer from astdb: '540' at 540@192.168.1.62:5060 for 3600
[Mar 10 12:01:14] DEBUG[15131]: db.c:200 ast_db_get: Unable to find key '597' in family 'SIP/Registry'
[Mar 10 12:01:14] DEBUG[15131]: db.c:200 ast_db_get: Unable to find key '598' in family 'SIP/Registry'
[Mar 10 12:01:14] DEBUG[15131]: db.c:200 ast_db_get: Unable to find key '599' in family 'SIP/Registry'
[Mar 10 12:01:14] DEBUG[15131]: db.c:200 ast_db_get: Unable to find key '1001' in family 'SIP/Registry'
[Mar 10 12:01:14] DEBUG[15131]: db.c:200 ast_db_get: Unable to find key '1002' in family 'SIP/Registry'
[Mar 10 12:01:14] DEBUG[15131]: db.c:200 ast_db_get: Unable to find key '1003' in family 'SIP/Registry'
[Mar 10 12:01:14] DEBUG[15131]: db.c:200 ast_db_get: Unable to find key '1004' in family 'SIP/Registry'
[Mar 10 12:01:14] DEBUG[15131]: db.c:200 ast_db_get: Unable to find key '521' in family 'SIP/Registry'

[Mar 10 12:01:14]     -- SIP Seeding peer from astdb: '522' at 522@192.168.1.95:5060 for 60

[Mar 10 12:01:14]     -- SIP Seeding peer from astdb: '523' at 523@192.168.1.85:5060 for 60

[Mar 10 12:01:14]     -- SIP Seeding peer from astdb: '524' at 524@192.168.1.61:5060 for 60

[Mar 10 12:01:14]     -- SIP Seeding peer from astdb: '525' at 525@192.168.1.76:5060 for 60

[Mar 10 12:01:14]     -- SIP Seeding peer from astdb: '526' at 526@192.168.1.66:5060 for 60

[Mar 10 12:01:14]     -- SIP Seeding peer from astdb: '528' at 528@192.168.1.93:5060 for 60

[Mar 10 12:01:14]     -- SIP Seeding peer from astdb: '529' at 529@192.168.1.98:5060 for 60
[Mar 10 12:01:14] WARNING[15131]: frame.c:1030 ast_parse_allow_disallow: Cannot allow unknown format 'h264'
[Mar 10 12:01:14] WARNING[15131]: frame.c:1030 ast_parse_allow_disallow: Cannot allow unknown format 'h264'
[Mar 10 12:01:14] WARNING[15131]: frame.c:1030 ast_parse_allow_disallow: Cannot allow unknown format 'h264'
[Mar 10 12:01:14] WARNING[15131]: frame.c:1030 ast_parse_allow_disallow: Cannot allow unknown format 'h264'

[Mar 10 12:01:14]     -- SIP Seeding peer from astdb: '541' at 541@192.168.1.90:5060 for 3600
[Mar 10 12:01:14] DEBUG[15131]: db.c:200 ast_db_get: Unable to find key '550' in family 'SIP/Registry'

[Mar 10 12:01:14]     -- SIP Seeding peer from astdb: '551' at 551@192.168.1.97:5060 for 60

[Mar 10 12:01:14]     -- SIP Seeding peer from astdb: '592' at 592@68.58.36.157:5060 for 3600

[Mar 10 12:01:14]     -- SIP Seeding peer from astdb: '593' at 593@192.168.1.83:5060 for 3600
[Mar 10 12:01:14] DEBUG[15131]: db.c:200 ast_db_get: Unable to find key '594' in family 'SIP/Registry'

[Mar 10 12:01:14]     -- SIP Seeding peer from astdb: '595' at 595@192.168.1.86:5060 for 3600

[Mar 10 12:01:14]     -- SIP Seeding peer from astdb: '596' at 596@192.168.1.86:5062 for 3600

[Mar 10 12:01:14]   == SIP Listening on 24.123.23.170:5060

[Mar 10 12:01:14]   == Using TOS bits 0

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/sip_notify.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/sip_notify.conf
[Mar 10 12:01:14] Found
[Mar 10 12:01:14] DEBUG[15131]: channel.c:344 ast_channel_register: Registered handler for 'SIP' (Session Initiation Protocol (SIP))

[Mar 10 12:01:14]   == Registered channel type 'SIP' (Session Initiation Protocol (SIP))

[Mar 10 12:01:14]   == Registered application 'SIPDtmfMode'

[Mar 10 12:01:14]   == Registered application 'SIPAddHeader'

[Mar 10 12:01:14]   == Registered application 'SIPGetHeader'

[Mar 10 12:01:14]   == Registered custom function SIP_HEADER

[Mar 10 12:01:14]   == Registered custom function SIPPEER

[Mar 10 12:01:14]   == Registered custom function SIPCHANINFO

[Mar 10 12:01:14]   == Registered custom function CHECKSIPDOMAIN

[Mar 10 12:01:14]   == Manager registered action SIPpeers

[Mar 10 12:01:14]   == Manager registered action SIPshowpeer
[Mar 10 12:01:14] DEBUG[15131]: chan_sip.c:3168 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP)

[Mar 10 12:01:14]  [chan_agent.so][Mar 10 12:01:14]  => (Agent Proxy Channel)
[Mar 10 12:01:14] DEBUG[15131]: channel.c:344 ast_channel_register: Registered handler for 'Agent' (Call Agent Proxy Channel)

[Mar 10 12:01:14]   == Registered channel type 'Agent' (Call Agent Proxy Channel)

[Mar 10 12:01:14]   == Registered application 'AgentLogin'

[Mar 10 12:01:14]   == Registered application 'AgentCallbackLogin'

[Mar 10 12:01:14]   == Registered application 'AgentMonitorOutgoing'

[Mar 10 12:01:14]   == Manager registered action Agents

[Mar 10 12:01:14]   == Manager registered action AgentLogoff

[Mar 10 12:01:14]   == Manager registered action AgentCallbackLogin

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/agents.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/agents.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]  [chan_local.so][Mar 10 12:01:14]  => (Local Proxy Channel)
[Mar 10 12:01:14] DEBUG[15131]: channel.c:344 ast_channel_register: Registered handler for 'Local' (Local Proxy Channel Driver)

[Mar 10 12:01:14]   == Registered channel type 'Local' (Local Proxy Channel Driver)

[Mar 10 12:01:14]  [chan_zap.so][Mar 10 12:01:14]  => (Zapata Telephony w/PRI)

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/zapata.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/zapata.conf
[Mar 10 12:01:14] Found
[Mar 10 12:01:14] DEBUG[15131]: chan_zap.c:1372 update_conf: Updated conferencing on 1, with 0 conference users

[Mar 10 12:01:14]     -- Registered channel 1, FXS Kewlstart signalling
[Mar 10 12:01:14] DEBUG[15131]: chan_zap.c:1372 update_conf: Updated conferencing on 2, with 0 conference users

[Mar 10 12:01:14]     -- Registered channel 2, FXS Kewlstart signalling
[Mar 10 12:01:14] DEBUG[15131]: chan_zap.c:1372 update_conf: Updated conferencing on 3, with 0 conference users

[Mar 10 12:01:14]     -- Registered channel 3, FXS Kewlstart signalling
[Mar 10 12:01:14] DEBUG[15131]: chan_zap.c:1372 update_conf: Updated conferencing on 4, with 0 conference users

[Mar 10 12:01:14]     -- Registered channel 4, FXS Kewlstart signalling
[Mar 10 12:01:14] DEBUG[15131]: chan_zap.c:1372 update_conf: Updated conferencing on 5, with 0 conference users

[Mar 10 12:01:14]     -- Registered channel 5, FXS Kewlstart signalling
[Mar 10 12:01:14] DEBUG[15131]: chan_zap.c:1372 update_conf: Updated conferencing on 6, with 0 conference users

[Mar 10 12:01:14]     -- Registered channel 6, FXS Kewlstart signalling
[Mar 10 12:01:14] DEBUG[15131]: chan_zap.c:1372 update_conf: Updated conferencing on 7, with 0 conference users

[Mar 10 12:01:14]     -- Registered channel 7, FXS Kewlstart signalling
[Mar 10 12:01:14] DEBUG[15131]: chan_zap.c:1372 update_conf: Updated conferencing on 8, with 0 conference users

[Mar 10 12:01:14]     -- Registered channel 8, FXS Kewlstart signalling

[Mar 10 12:01:14]     -- Automatically generated pseudo channel
[Mar 10 12:01:14] DEBUG[15131]: channel.c:344 ast_channel_register: Registered handler for 'Zap' (Zapata Telephony Driver w/PRI)

[Mar 10 12:01:14]   == Registered channel type 'Zap' (Zapata Telephony Driver w/PRI)

[Mar 10 12:01:14]   == Manager registered action ZapTransfer

[Mar 10 12:01:14]   == Manager registered action ZapHangup

[Mar 10 12:01:14]   == Manager registered action ZapDialOffhook

[Mar 10 12:01:14]   == Manager registered action ZapDNDon

[Mar 10 12:01:14]   == Manager registered action ZapDNDoff

[Mar 10 12:01:14]   == Manager registered action ZapShowChannels

[Mar 10 12:01:14]  [chan_oss.so][Mar 10 12:01:14]  => (OSS Console Channel Driver)

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/oss.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/oss.conf
[Mar 10 12:01:14] Found
[Mar 10 12:01:14] DEBUG[15131]: channel.c:344 ast_channel_register: Registered handler for 'Console' (OSS Console Channel Driver)

[Mar 10 12:01:14]   == Registered channel type 'Console' (OSS Console Channel Driver)

[Mar 10 12:01:14]  [app_setcdruserfield.so][Mar 10 12:01:14]  => (CDR user field apps)

[Mar 10 12:01:14]   == Registered application 'SetCDRUserField'

[Mar 10 12:01:14]   == Registered application 'AppendCDRUserField'

[Mar 10 12:01:14]   == Manager registered action SetCDRUserField

[Mar 10 12:01:14]  [app_queue.so][Mar 10 12:01:14]  => (True Call Queueing)

[Mar 10 12:01:14]   == Registered application 'Queue'

[Mar 10 12:01:14]   == Manager registered action Queues

[Mar 10 12:01:14]   == Manager registered action QueueStatus

[Mar 10 12:01:14]   == Manager registered action QueueAdd

[Mar 10 12:01:14]   == Manager registered action QueueRemove

[Mar 10 12:01:14]   == Manager registered action QueuePause

[Mar 10 12:01:14]   == Registered application 'AddQueueMember'

[Mar 10 12:01:14]   == Registered application 'RemoveQueueMember'

[Mar 10 12:01:14]   == Registered application 'PauseQueueMember'

[Mar 10 12:01:14]   == Registered application 'UnpauseQueueMember'

[Mar 10 12:01:14]   == Registered custom function QUEUEAGENTCOUNT

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/queues.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/queues.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]  [app_dictate.so][Mar 10 12:01:14]  => (Virtual Dictation Machine)

[Mar 10 12:01:14]   == Registered application 'Dictate'

[Mar 10 12:01:14]  [app_disa.so][Mar 10 12:01:14]  => (DISA (Direct Inward System Access) Application)

[Mar 10 12:01:14]   == Registered application 'DISA'

[Mar 10 12:01:14]  [app_chanisavail.so][Mar 10 12:01:14]  => (Check channel availability)

[Mar 10 12:01:14]   == Registered application 'ChanIsAvail'

[Mar 10 12:01:14]  [cdr_custom.so][Mar 10 12:01:14]  => (Customizable Comma Separated Values CDR Backend)

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/cdr_custom.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/cdr_custom.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]  [codec_alaw.so][Mar 10 12:01:14]  => (A-law Coder/Decoder)

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/codecs.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/codecs.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]     -- codec_alaw: using generic PLC

[Mar 10 12:01:14]   == Registered translator 'alawtolin' from format alaw to slin, cost 1
[Mar 10 12:01:14] DEBUG[15131]: translate.c:278 rebuild_matrix: Resetting translation matrix

[Mar 10 12:01:14]   == Registered translator 'lintoalaw' from format slin to alaw, cost 1
[Mar 10 12:01:14] DEBUG[15131]: translate.c:278 rebuild_matrix: Resetting translation matrix

[Mar 10 12:01:14]  [app_txtcidname.so][Mar 10 12:01:14]  => (TXTCIDName)

[Mar 10 12:01:14]   == Registered application 'TXTCIDName'

[Mar 10 12:01:14]  [format_ogg_vorbis.so][Mar 10 12:01:14]  => (OGG/Vorbis audio)

[Mar 10 12:01:14]   == Registered file format ogg_vorbis, extension(s) ogg

[Mar 10 12:01:14]  [app_zapras.so][Mar 10 12:01:14]  => (Zap RAS Application)

[Mar 10 12:01:14]   == Registered application 'ZapRAS'

[Mar 10 12:01:14]  [format_pcm.so][Mar 10 12:01:14]  => (Raw uLaw 8khz Audio support (PCM))

[Mar 10 12:01:14]   == Registered file format pcm, extension(s) pcm|ulaw|ul|mu

[Mar 10 12:01:14]  [format_g726.so][Mar 10 12:01:14]  => (Raw G.726 (16/24/32/40kbps) data)

[Mar 10 12:01:14]   == Registered file format g726-40, extension(s) g726-40

[Mar 10 12:01:14]   == Registered file format g726-32, extension(s) g726-32

[Mar 10 12:01:14]   == Registered file format g726-24, extension(s) g726-24

[Mar 10 12:01:14]   == Registered file format g726-16, extension(s) g726-16

[Mar 10 12:01:14]  [app_page.so][Mar 10 12:01:14]  => (Page Multiple Phones)

[Mar 10 12:01:14]   == Registered application 'Page'

[Mar 10 12:01:14]  [app_random.so][Mar 10 12:01:14]  => (Random goto)

[Mar 10 12:01:14]   == Registered application 'Random'

[Mar 10 12:01:14]  [format_gsm.so][Mar 10 12:01:14]  => (Raw GSM data)

[Mar 10 12:01:14]   == Registered file format gsm, extension(s) gsm

[Mar 10 12:01:14]  [codec_ilbc.so][Mar 10 12:01:14]  => (iLBC/PCM16 (signed linear) Codec Translator)

[Mar 10 12:01:14]   == Registered translator 'ilbctolin' from format ilbc to slin, cost 2
[Mar 10 12:01:14] DEBUG[15131]: translate.c:278 rebuild_matrix: Resetting translation matrix
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6
[Mar 10 12:01:14] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 77bc05f513d387964ce8856153eba1df@24.123.23.170 Their Tag  Our tag: as2b70d731
[Mar 10 12:01:14] DEBUG[15149]: chan_sip.c:3168 sip_alloc: Allocating new SIP dialog for 55ae6647666e@24.123.23.170 - REGISTER (No RTP)
[Mar 10 12:01:14] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER
[Mar 10 12:01:14] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae6647666e@24.123.23.170 Their Tag 1c28ba8194fa9bb96cd0ff11cef75bd0 Our tag: as5b6350dc
[Mar 10 12:01:14] DEBUG[15149]: chan_sip.c:3216 find_call: = Found Their Call ID: 77bc05f513d387964ce8856153eba1df@24.123.23.170 Their Tag  Our tag: as2b70d731
[Mar 10 12:01:14] DEBUG[15149]: chan_sip.c:1412 __sip_ack: Stopping retransmission on '77bc05f513d387964ce8856153eba1df@24.123.23.170' of Request 102: Match Found
[Mar 10 12:01:14] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer Broadvoice
[Mar 10 12:01:14] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/Broadvoice - state 1 (Not in use)
[Mar 10 12:01:14] DEBUG[15152]: app_queue.c:500 changethread: Device 'SIP/Broadvoice' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.

[Mar 10 12:01:14]   == Registered translator 'lintoilbc' from format slin to ilbc, cost 13
[Mar 10 12:01:14] DEBUG[15131]: translate.c:278 rebuild_matrix: Resetting translation matrix
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from unknown to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6

[Mar 10 12:01:14]  [format_g723.so][Mar 10 12:01:14]  => (G.723.1 Simple Timestamp File Format)

[Mar 10 12:01:14]   == Registered file format g723sf, extension(s) g723|g723sf

[Mar 10 12:01:14]  [format_g729.so][Mar 10 12:01:14]  => (Raw G729 data)

[Mar 10 12:01:14]   == Registered file format g729, extension(s) g729

[Mar 10 12:01:14]  [cdr_csv.so][Mar 10 12:01:14]  => (Comma Separated Values CDR Backend)

[Mar 10 12:01:14]  [app_softhangup.so][Mar 10 12:01:14]  => (Hangs up the requested channel)

[Mar 10 12:01:14]   == Registered application 'SoftHangup'

[Mar 10 12:01:14]  [app_parkandannounce.so][Mar 10 12:01:14]  => (Call Parking and Announce Application)

[Mar 10 12:01:14]   == Registered application 'ParkAndAnnounce'

[Mar 10 12:01:14]  [app_read.so][Mar 10 12:01:14]  => (Read Variable Application)

[Mar 10 12:01:14]   == Registered application 'Read'

[Mar 10 12:01:14]  [app_hasnewvoicemail.so][Mar 10 12:01:14]  => (Indicator for whether a voice mailbox has messages in a given folder.)

[Mar 10 12:01:14]   == Registered custom function VMCOUNT

[Mar 10 12:01:14]   == Registered application 'HasVoicemail'

[Mar 10 12:01:14]   == Registered application 'HasNewVoicemail'

[Mar 10 12:01:14]  [app_privacy.so][Mar 10 12:01:14]  => (Require phone number to be entered, if no CallerID sent)

[Mar 10 12:01:14]   == Registered application 'PrivacyManager'

[Mar 10 12:01:14]  [app_alarmreceiver.so][Mar 10 12:01:14]  => (Alarm Receiver for Asterisk)

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/alarmreceiver.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/alarmreceiver.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]   == Registered application 'AlarmReceiver'

[Mar 10 12:01:14]  [format_sln.so][Mar 10 12:01:14]  => (Raw Signed Linear Audio support (SLN))

[Mar 10 12:01:14]   == Registered file format sln, extension(s) sln|raw

[Mar 10 12:01:14]  [app_waitforsilence.so][Mar 10 12:01:14]  => (Wait For Silence)

[Mar 10 12:01:14]   == Registered application 'WaitForSilence'

[Mar 10 12:01:14]  [app_setcidnum.so][Mar 10 12:01:14]  => (Set CallerID Number)

[Mar 10 12:01:14]   == Registered application 'SetCIDNum'

[Mar 10 12:01:14]  [app_url.so][Mar 10 12:01:14]  => (Send URL Applications)

[Mar 10 12:01:14]   == Registered application 'SendURL'

[Mar 10 12:01:14]  [app_festival.so][Mar 10 12:01:14]  => (Simple Festival Interface)

[Mar 10 12:01:14]   == Registered application 'Festival'

[Mar 10 12:01:14]  [app_meetme.so][Mar 10 12:01:14]  => (MeetMe conference bridge)

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/meetme.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/meetme.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/express.demonstration.meetme.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/express.demonstration.meetme.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]   == Registered application 'MeetMeAdmin'

[Mar 10 12:01:14]   == Registered application 'MeetMeCount'

[Mar 10 12:01:14]   == Registered application 'MeetMe'

[Mar 10 12:01:14]  [app_groupcount.so][Mar 10 12:01:14]  => (Group Management Routines)

[Mar 10 12:01:14]   == Registered application 'GetGroupCount'

[Mar 10 12:01:14]   == Registered application 'SetGroup'

[Mar 10 12:01:14]   == Registered application 'CheckGroup'

[Mar 10 12:01:14]   == Registered application 'GetGroupMatchCount'

[Mar 10 12:01:14]  [app_md5.so][Mar 10 12:01:14]  => (MD5 checksum applications)

[Mar 10 12:01:14]   == Registered application 'MD5Check'

[Mar 10 12:01:14]   == Registered application 'MD5'

[Mar 10 12:01:14]  [app_mixmonitor.so][Mar 10 12:01:14]  => (Mixed Audio Monitoring Application)

[Mar 10 12:01:14]   == Registered application 'MixMonitor'

[Mar 10 12:01:14]  [app_ices.so][Mar 10 12:01:14]  => (Encode and Stream via icecast and ices)

[Mar 10 12:01:14]   == Registered application 'ICES'

[Mar 10 12:01:14]  [format_jpeg.so][Mar 10 12:01:14]  => (JPEG (Joint Picture Experts Group) Image Format)

[Mar 10 12:01:14]   == Registered format 'jpg' (JPEG (Joint Picture Experts Group))

[Mar 10 12:01:14]  [app_setrdnis.so][Mar 10 12:01:14]  => (Set RDNIS Number)

[Mar 10 12:01:14]   == Registered application 'SetRDNIS'

[Mar 10 12:01:14]  [skipping chan_alsa.so]

[Mar 10 12:01:14]  [app_stack.so][Mar 10 12:01:14]  => (Stack Routines)

[Mar 10 12:01:14]   == Registered application 'StackPop'

[Mar 10 12:01:14]   == Registered application 'Return'

[Mar 10 12:01:14]   == Registered application 'GosubIf'

[Mar 10 12:01:14]   == Registered application 'Gosub'

[Mar 10 12:01:14]  [format_pcm_alaw.so][Mar 10 12:01:14]  => (Raw aLaw 8khz PCM Audio support)

[Mar 10 12:01:14]   == Registered file format alaw, extension(s) alaw|al

[Mar 10 12:01:14]  [app_macro.so][Mar 10 12:01:14]  => (Extension Macros)

[Mar 10 12:01:14]   == Registered application 'MacroExit'

[Mar 10 12:01:14]   == Registered application 'MacroIf'

[Mar 10 12:01:14]   == Registered application 'Macro'

[Mar 10 12:01:14]  [app_rxfax.so][Mar 10 12:01:14]  => (Trivial FAX Receive Application)

[Mar 10 12:01:14]   == Registered application 'RxFAX'

[Mar 10 12:01:14]  [app_realtime.so][Mar 10 12:01:14]  => (Realtime Data Lookup/Rewrite)

[Mar 10 12:01:14]   == Registered application 'RealTimeUpdate'

[Mar 10 12:01:14]   == Registered application 'RealTime'

[Mar 10 12:01:14]  [app_db.so][Mar 10 12:01:14]  => (Database Access Functions)

[Mar 10 12:01:14]   == Registered application 'DBget'

[Mar 10 12:01:14]   == Registered application 'DBput'

[Mar 10 12:01:14]   == Registered application 'DBdel'

[Mar 10 12:01:14]   == Registered application 'DBdeltree'

[Mar 10 12:01:14]  [app_milliwatt.so][Mar 10 12:01:14]  => (Digital Milliwatt (mu-law) Test Application)

[Mar 10 12:01:14]   == Registered application 'Milliwatt'

[Mar 10 12:01:14]  [func_enum.so][Mar 10 12:01:14]  => (ENUM Related Functions)

[Mar 10 12:01:14]   == Registered custom function ENUMLOOKUP

[Mar 10 12:01:14]   == Registered custom function TXTCIDNAME

[Mar 10 12:01:14]  [app_dumpchan.so][Mar 10 12:01:14]  => (Dump Info About The Calling Channel)

[Mar 10 12:01:14]   == Registered application 'DumpChan'

[Mar 10 12:01:14]  [app_setcidname.so][Mar 10 12:01:14]  => (Set CallerID Name)

[Mar 10 12:01:14]   == Registered application 'SetCIDName'

[Mar 10 12:01:14]  [app_test.so][Mar 10 12:01:14]  => (Interface Test Application)

[Mar 10 12:01:14]   == Registered application 'TestClient'

[Mar 10 12:01:14]   == Registered application 'TestServer'

[Mar 10 12:01:14]  [app_verbose.so][Mar 10 12:01:14]  => (Send verbose output)

[Mar 10 12:01:14]   == Registered application 'Verbose'

[Mar 10 12:01:14]  [cdr_manager.so][Mar 10 12:01:14]  => (Asterisk Call Manager CDR Backend)

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/cdr_manager.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/cdr_manager.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]  [app_sendtext.so][Mar 10 12:01:14]  => (Send Text Applications)

[Mar 10 12:01:14]   == Registered application 'SendText'

[Mar 10 12:01:14]  [codec_lpc10.so][Mar 10 12:01:14]  => (LPC10 2.4kbps (signed linear) Voice Coder)

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/codecs.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/codecs.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]     -- codec_lpc10: using generic PLC

[Mar 10 12:01:14]   == Registered translator 'lpc10tolin' from format lpc10 to slin, cost 1
[Mar 10 12:01:14] DEBUG[15131]: translate.c:278 rebuild_matrix: Resetting translation matrix
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from unknown to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from unknown to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6

[Mar 10 12:01:14]   == Registered translator 'lintolpc10' from format slin to lpc10, cost 3
[Mar 10 12:01:14] DEBUG[15131]: translate.c:278 rebuild_matrix: Resetting translation matrix
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from unknown to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from unknown to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6

[Mar 10 12:01:14]  [app_echo.so][Mar 10 12:01:14]  => (Simple Echo Application)

[Mar 10 12:01:14]   == Registered application 'Echo'

[Mar 10 12:01:14]  [app_record.so][Mar 10 12:01:14]  => (Trivial Record Application)

[Mar 10 12:01:14]   == Registered application 'Record'

[Mar 10 12:01:14]  [app_zapbarge.so][Mar 10 12:01:14]  => (Barge in on Zap channel application)

[Mar 10 12:01:14]   == Registered application 'ZapBarge'

[Mar 10 12:01:14]  [codec_gsm.so][Mar 10 12:01:14]  => (GSM/PCM16 (signed linear) Codec Translator)

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/codecs.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/codecs.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]     -- codec_gsm: using generic PLC

[Mar 10 12:01:14]   == Registered translator 'gsmtolin' from format gsm to slin, cost 1
[Mar 10 12:01:14] DEBUG[15131]: translate.c:278 rebuild_matrix: Resetting translation matrix
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from g723 to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from unknown to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from unknown to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6

[Mar 10 12:01:14]   == Registered translator 'lintogsm' from format slin to gsm, cost 2
[Mar 10 12:01:14] DEBUG[15131]: translate.c:278 rebuild_matrix: Resetting translation matrix
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from g723 to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from unknown to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from unknown to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6

[Mar 10 12:01:14]  [app_math.so][Mar 10 12:01:14]  => (Basic Math Functions)

[Mar 10 12:01:14]   == Registered application 'Math'

[Mar 10 12:01:14]  [app_dial.so][Mar 10 12:01:14]  => (Dialing Application)

[Mar 10 12:01:14]   == Registered application 'Dial'

[Mar 10 12:01:14]   == Registered application 'RetryDial'

[Mar 10 12:01:14]  [app_adsiprog.so][Mar 10 12:01:14]  => (Asterisk ADSI Programming Application)

[Mar 10 12:01:14]   == Registered application 'ADSIProg'

[Mar 10 12:01:14]  [app_curl.so][Mar 10 12:01:14]  => (Load external URL)

[Mar 10 12:01:14]   == Registered custom function CURL

[Mar 10 12:01:14]   == Registered application 'Curl'

[Mar 10 12:01:14]  [app_exec.so][Mar 10 12:01:14]  => (Executes applications)

[Mar 10 12:01:14]   == Registered application 'Exec'

[Mar 10 12:01:14]  [format_wav_gsm.so][Mar 10 12:01:14]  => (Microsoft WAV format (Proprietary GSM))

[Mar 10 12:01:14]   == Registered file format wav49, extension(s) WAV|wav49

[Mar 10 12:01:14]  [app_lookupcidname.so][Mar 10 12:01:14]  => (Look up CallerID Name from local database)

[Mar 10 12:01:14]   == Registered application 'LookupCIDName'

[Mar 10 12:01:14]  [app_txfax.so][Mar 10 12:01:14]  => (Trivial FAX Transmit Application)

[Mar 10 12:01:14]   == Registered application 'TxFAX'

[Mar 10 12:01:14]  [app_directory.so][Mar 10 12:01:14]  => (Extension Directory)

[Mar 10 12:01:14]   == Registered application 'Directory'

[Mar 10 12:01:14]  [app_flash.so][Mar 10 12:01:14]  => (Flash zap trunk application)

[Mar 10 12:01:14]   == Registered application 'Flash'

[Mar 10 12:01:14]  [app_zapscan.so][Mar 10 12:01:14]  => (Scan Zap channels application)

[Mar 10 12:01:14]   == Registered application 'ZapScan'

[Mar 10 12:01:14]  [format_ilbc.so][Mar 10 12:01:14]  => (Raw iLBC data)

[Mar 10 12:01:14]   == Registered file format iLBC, extension(s) ilbc

[Mar 10 12:01:14]  [app_setcallerid.so][Mar 10 12:01:14]  => (Set CallerID Application)

[Mar 10 12:01:14]   == Registered application 'SetCallerPres'

[Mar 10 12:01:14]   == Registered application 'SetCallerID'

[Mar 10 12:01:14]  [app_transfer.so][Mar 10 12:01:14]  => (Transfer)

[Mar 10 12:01:14]   == Registered application 'Transfer'

[Mar 10 12:01:14]  [codec_g726.so][Mar 10 12:01:14]  => (ITU G.726-32kbps G726 Transcoder)

[Mar 10 12:01:14]   == Parsing '/etc/asterisk/codecs.conf': [Mar 10 12:01:14] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/codecs.conf
[Mar 10 12:01:14] Found

[Mar 10 12:01:14]     -- codec_g726: using generic PLC

[Mar 10 12:01:14]   == Registered translator 'g726tolin' from format g726 to slin, cost 1
[Mar 10 12:01:14] DEBUG[15131]: translate.c:278 rebuild_matrix: Resetting translation matrix
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from g723 to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from unknown to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from ulaw to g723, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from ulaw to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from ulaw to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from ulaw to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from unknown to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6

[Mar 10 12:01:14]   == Registered translator 'lintog726' from format slin to g726, cost 1
[Mar 10 12:01:14] DEBUG[15131]: translate.c:278 rebuild_matrix: Resetting translation matrix
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from g723 to ulaw, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from g723 to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to ulaw, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6
[Mar 10 12:01:14] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from unknown to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from ulaw to g723, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from ulaw to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from ulaw to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from ulaw to unknown, via 6
[Mar 10 12:01:16] DEBUG[15149]: chan_sip.c:3168 sip_alloc: Allocating new SIP dialog for 71f439ca643c96aa63da468e6d077ab4@24.123.23.170 - REGISTER (No RTP)
[Mar 10 12:01:16] DEBUG[15136]: chan_iax2.c:9448 iax2_devicestate: Checking device state for device 614
[Mar 10 12:01:16] DEBUG[15136]: chan_iax2.c:9456 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 614? addr=-1493063488, defaddr=0 maxms=0, lastms=0
[Mar 10 12:01:16] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for IAX2/614 - state 1 (Not in use)
[Mar 10 12:01:16] DEBUG[15136]: chan_iax2.c:9448 iax2_devicestate: Checking device state for device 614
[Mar 10 12:01:16] DEBUG[15136]: chan_iax2.c:9456 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 614? addr=-1493063488, defaddr=0 maxms=0, lastms=0
[Mar 10 12:01:16] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for IAX2/614 - state 1 (Not in use)
[Mar 10 12:01:16] DEBUG[15148]: db.c:200 ast_db_get: Unable to find key 'si-000fd300002e' in family 'iax/provisioning/cache'
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to ulaw, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from unknown to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6

[Mar 10 12:01:16]  [format_au.so][Mar 10 12:01:16]  => (Sun Microsystems AU format (signed linear))

[Mar 10 12:01:16]   == Registered file format au, extension(s) au

[Mar 10 12:01:16]  [app_controlplayback.so][Mar 10 12:01:16]  => (Control Playback Application)

[Mar 10 12:01:16]   == Registered application 'ControlPlayback'

[Mar 10 12:01:16]  [func_uri.so][Mar 10 12:01:16]  => (URI encode/decode functions)

[Mar 10 12:01:16]   == Registered custom function URIDECODE

[Mar 10 12:01:16]   == Registered custom function URIENCODE

[Mar 10 12:01:16]  [app_chanspy.so][Mar 10 12:01:16]  => (Listen to the audio of an active channel
)

[Mar 10 12:01:16]   == Registered application 'ChanSpy'

[Mar 10 12:01:16]  [app_enumlookup.so][Mar 10 12:01:16]  => (ENUM Lookup)

[Mar 10 12:01:16]   == Registered application 'EnumLookup'

[Mar 10 12:01:16]   == Parsing '/etc/asterisk/enum.conf': [Mar 10 12:01:16] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/enum.conf
[Mar 10 12:01:16] Found

[Mar 10 12:01:16]  [app_settransfercapability.so][Mar 10 12:01:16]  => (Set ISDN Transfer Capability)

[Mar 10 12:01:16]   == Registered application 'SetTransferCapability'

[Mar 10 12:01:16]  [app_directed_pickup.so][Mar 10 12:01:16]  => (Directed Call Pickup Application)

[Mar 10 12:01:16]   == Registered application 'Pickup'

[Mar 10 12:01:16]  [app_nbscat.so][Mar 10 12:01:16]  => (Silly NBS Stream Application)

[Mar 10 12:01:16]   == Registered application 'NBScat'

[Mar 10 12:01:16]  [func_callerid.so][Mar 10 12:01:16]  => (Caller ID related dialplan function)

[Mar 10 12:01:16]   == Registered custom function CALLERID

[Mar 10 12:01:16]  [app_voicemail.so][Mar 10 12:01:16]  => (Comedian Mail (Voicemail System))

[Mar 10 12:01:16]   == Registered application 'VoiceMail'

[Mar 10 12:01:16]   == Registered application 'VoiceMailMain'

[Mar 10 12:01:16]   == Registered application 'MailboxExists'

[Mar 10 12:01:16]   == Registered application 'VMAuthenticate'

[Mar 10 12:01:16]   == Parsing '/etc/asterisk/voicemail.conf': [Mar 10 12:01:16] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/voicemail.conf
[Mar 10 12:01:16] Found
[Mar 10 12:01:16] DEBUG[15131]: app_voicemail.c:6006 load_config: VM Review Option disabled globally
[Mar 10 12:01:16] DEBUG[15131]: app_voicemail.c:6012 load_config: VM Operator break disabled globally
[Mar 10 12:01:16] DEBUG[15131]: app_voicemail.c:6018 load_config: VM CID Info before msg disabled globally
[Mar 10 12:01:16] DEBUG[15131]: app_voicemail.c:6030 load_config: ENVELOPE before msg enabled globally
[Mar 10 12:01:16] DEBUG[15131]: app_voicemail.c:6036 load_config: Duration info before msg enabled globally
[Mar 10 12:01:16] DEBUG[15131]: app_voicemail.c:6051 load_config: We are not going to skip to the next msg after save/delete

[Mar 10 12:01:16]  [app_externalivr.so][Mar 10 12:01:16]  => (External IVR Interface Application)

[Mar 10 12:01:16]   == Registered application 'ExternalIVR'

[Mar 10 12:01:16]  [app_mp3.so][Mar 10 12:01:16] DEBUG[15158]: app_queue.c:500 changethread: Device 'IAX2/614' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:01:16] DEBUG[15157]: app_queue.c:500 changethread: [Mar 10 12:01:16]  => (Silly MP3 Application)

[Mar 10 12:01:16]   == Registered application 'MP3Player'

[Mar 10 12:01:16]  [app_readfile.so][Mar 10 12:01:16]  => (Stores output of file into a variable)

[Mar 10 12:01:16]   == Registered application 'ReadFile'

[Mar 10 12:01:16]  [app_image.so][Mar 10 12:01:16]  => (Image Transmission Application)

[Mar 10 12:01:16]   == Registered application 'SendImage'

[Mar 10 12:01:16]  [app_while.so][Mar 10 12:01:16]  => (While Loops and Conditional Execution)

[Mar 10 12:01:16]   == Registered application 'While'

[Mar 10 12:01:16]   == Registered application 'ExecIf'

[Mar 10 12:01:16]   == Registered application 'EndWhile'

[Mar 10 12:01:16]  [app_sayunixtime.so]Device 'IAX2/614' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:01:16]  => (Say time)

[Mar 10 12:01:16]   == Registered application 'SayUnixTime'

[Mar 10 12:01:16]   == Registered application 'DateTime'

[Mar 10 12:01:16]  [app_waitforring.so][Mar 10 12:01:16]  => (Waits until first ring after time)

[Mar 10 12:01:16]   == Registered application 'WaitForRing'

[Mar 10 12:01:16]  [app_zapateller.so][Mar 10 12:01:16]  => (Block Telemarketers with Special Information Tone)

[Mar 10 12:01:16]   == Registered application 'Zapateller'

[Mar 10 12:01:16]  [app_cdr.so][Mar 10 12:01:16]  => (Tell Asterisk to not maintain a CDR for the current call)

[Mar 10 12:01:16]   == Registered application 'NoCDR'

[Mar 10 12:01:16]  [app_playback.so][Mar 10 12:01:16]  => (Sound File Playback Application)

[Mar 10 12:01:16]   == Registered application 'Playback'

[Mar 10 12:01:16]  [app_senddtmf.so][Mar 10 12:01:16]  => (Send DTMF digits Application)

[Mar 10 12:01:16]   == Registered application 'SendDTMF'

[Mar 10 12:01:16]  [app_forkcdr.so][Mar 10 12:01:16]  => (Fork The CDR into 2 separate entities.)

[Mar 10 12:01:16]   == Registered application 'ForkCDR'

[Mar 10 12:01:16]  [app_userevent.so][Mar 10 12:01:16]  => (Custom User Event Application)

[Mar 10 12:01:16]   == Registered application 'UserEvent'

[Mar 10 12:01:16]  [app_system.so][Mar 10 12:01:16]  => (Generic System() application)

[Mar 10 12:01:16]   == Registered application 'TrySystem'

[Mar 10 12:01:16]   == Registered application 'System'

[Mar 10 12:01:16]  [app_cut.so][Mar 10 12:01:16]  => (Cut out information from a string)

[Mar 10 12:01:16]   == Registered custom function CUT

[Mar 10 12:01:16]   == Registered custom function SORT

[Mar 10 12:01:16]   == Registered application 'Sort'

[Mar 10 12:01:16]   == Registered application 'Cut'

[Mar 10 12:01:16]  [codec_a_mu.so][Mar 10 12:01:16]  => (A-law and Mulaw direct Coder/Decoder)

[Mar 10 12:01:16]   == Registered translator 'alawtoulaw' from format alaw to ulaw, cost 1
[Mar 10 12:01:16] DEBUG[15131]: translate.c:278 rebuild_matrix: Resetting translation matrix
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from g723 to ulaw, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from g723 to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to ulaw, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from unknown to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from ulaw to g723, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from ulaw to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from ulaw to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from ulaw to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to gsm, via 3
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to gsm, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to ulaw, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from unknown to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from unknown to gsm, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6
[Mar 10 12:01:16] DEBUG[15149]: chan_sip.c:5564 transmit_register: Scheduled a registration timeout for sip.broadvoice.com id  #46 

[Mar 10 12:01:16] REGISTER attempt 1 to 3173241052@sip.broadvoice.com
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from g723 to gsm, via 3
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 6 cost path from ulaw to gsm, via 1
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from ulaw to gsm, via 3
[Mar 10 12:01:16] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 71f439ca643c96aa63da468e6d077ab4@24.123.23.170 Their Tag  Our tag: as19738910
[Mar 10 12:01:16] DEBUG[15149]: chan_sip.c:3216 find_call: = Found Their Call ID: 55ae6647666e@24.123.23.170 Their Tag 1c28ba8194fa9bb96cd0ff11cef75bd0 Our tag: as5b6350dc
[Mar 10 12:01:16] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:01:16]   == Registered translator 'ulawtoalaw' from format ulaw to alaw, cost 1
[Mar 10 12:01:16] DEBUG[15131]: translate.c:278 rebuild_matrix: Resetting translation matrix
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from g723 to ulaw, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from g723 to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from gsm to unknown, via 3
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from gsm to g723, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from gsm to ulaw, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 5 cost path from gsm to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 15 cost path from gsm to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 2
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from unknown to ulaw, via 2
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 6 cost path from unknown to unknown, via 2
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 2
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to ulaw, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from unknown to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from ulaw to g723, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from ulaw to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from ulaw to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from ulaw to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to gsm, via 3
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to gsm, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to ulaw, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from unknown to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from unknown to gsm, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from g723 to gsm, via 3
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 6 cost path from ulaw to gsm, via 1
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from ulaw to gsm, via 3

[Mar 10 12:01:16]  [format_h263.so][Mar 10 12:01:16]  => (Raw h263 data)

[Mar 10 12:01:16]   == Registered file format h263, extension(s) h263

[Mar 10 12:01:16]  [format_vox.so][Mar 10 12:01:16]  => (Dialogic VOX (ADPCM) File Format)

[Mar 10 12:01:16]   == Registered file format vox, extension(s) vox

[Mar 10 12:01:16]  [app_eval.so][Mar 10 12:01:16]  => (Reevaluates strings)

[Mar 10 12:01:16]   == Registered application 'Eval'

[Mar 10 12:01:16]  [app_talkdetect.so][Mar 10 12:01:16]  => (Playback with Talk Detection)

[Mar 10 12:01:16]   == Registered application 'BackgroundDetect'

[Mar 10 12:01:16]  [codec_adpcm.so][Mar 10 12:01:16]  => (Adaptive Differential PCM Coder/Decoder)

[Mar 10 12:01:16]   == Parsing '/etc/asterisk/codecs.conf': [Mar 10 12:01:16] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/codecs.conf
[Mar 10 12:01:16] Found

[Mar 10 12:01:16]     -- codec_adpcm: using generic PLC

[Mar 10 12:01:16]   == Registered translator 'adpcmtolin' from format adpcm to slin, cost 1
[Mar 10 12:01:16] DEBUG[15131]: translate.c:278 rebuild_matrix: Resetting translation matrix
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from g723 to ulaw, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from g723 to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from gsm to unknown, via 3
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from gsm to g723, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from gsm to ulaw, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 5 cost path from gsm to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 15 cost path from gsm to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 2
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from unknown to ulaw, via 2
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 6 cost path from unknown to unknown, via 2
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 2
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to ulaw, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from unknown to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from ulaw to g723, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from ulaw to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from ulaw to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from ulaw to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to ulaw, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from unknown to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to gsm, via 3
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to gsm, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to ulaw, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from unknown to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from unknown to gsm, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from g723 to gsm, via 3
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 6 cost path from ulaw to gsm, via 1
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from ulaw to gsm, via 3
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 6 cost path from unknown to gsm, via 1
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to gsm, via 3

[Mar 10 12:01:16]   == Registered translator 'lintoadpcm' from format slin to adpcm, cost 1
[Mar 10 12:01:16] DEBUG[15131]: translate.c:278 rebuild_matrix: Resetting translation matrix
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from g723 to ulaw, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from g723 to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from gsm to unknown, via 3
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from gsm to g723, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from gsm to ulaw, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from gsm to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 5 cost path from gsm to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 15 cost path from gsm to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 2
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from unknown to ulaw, via 2
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 2
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 6 cost path from unknown to unknown, via 2
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 2
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to ulaw, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from unknown to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from ulaw to g723, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from ulaw to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from ulaw to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from ulaw to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from ulaw to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to ulaw, via 6
[Mar 10 12:01:16] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from unknown to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to gsm, via 3
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to gsm, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to ulaw, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from unknown to unknown, via 6
[Mar 10 12:01:19] DEBUG[15148]: iax2-provision.c:249 iax_provision_version: Unable to create provisioning packet for 'si-000fd300002e'

[Mar 10 12:01:16]     -- Saved useragent "Uniden SIP Phone p2 Ver BS4.77" for peer 526
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:1209 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #47)) 
[Mar 10 12:01:19] DEBUG[15149]: sched.c:218 sched_settime: Request to schedule in the past?!?!
[Mar 10 12:01:19] DEBUG[15136]: chan_iax2.c:9448 iax2_devicestate: Checking device state for device 651
[Mar 10 12:01:19] DEBUG[15136]: chan_iax2.c:9456 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 651? addr=-1459509056, defaddr=0 maxms=0, lastms=0
[Mar 10 12:01:19] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for IAX2/651 - state 1 (Not in use)
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from unknown to gsm, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from g723 to gsm, via 3
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 6 cost path from ulaw to gsm, via 1
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from ulaw to gsm, via 3
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 6 cost path from unknown to gsm, via 1
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to gsm, via 3
[Mar 10 12:01:19] DEBUG[15160]: app_queue.c:500 changethread: Device 'IAX2/651' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.

[Mar 10 12:01:19]  [codec_ulaw.so][Mar 10 12:01:19]  => (Mu-law Coder/Decoder)

[Mar 10 12:01:19]   == Parsing '/etc/asterisk/codecs.conf': [Mar 10 12:01:19] DEBUG[15131]: config.c:595 config_text_file_load: Parsing /etc/asterisk/codecs.conf
[Mar 10 12:01:19] Found

[Mar 10 12:01:19]     -- codec_ulaw: using generic PLC

[Mar 10 12:01:19]   == Registered translator 'ulawtolin' from format ulaw to slin, cost 1
[Mar 10 12:01:19] DEBUG[15131]: translate.c:278 rebuild_matrix: Resetting translation matrix
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from g723 to ulaw, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from g723 to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from gsm to g723, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from gsm to ulaw, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from gsm to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from gsm to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from gsm to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 2
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 2
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 2
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 2
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 2
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to ulaw, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from unknown to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from ulaw to g723, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from ulaw to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from ulaw to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from ulaw to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from ulaw to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to ulaw, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from unknown to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to gsm, via 3
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to gsm, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to ulaw, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from unknown to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from unknown to gsm, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from g723 to gsm, via 3
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 6 cost path from ulaw to gsm, via 1
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from ulaw to gsm, via 3
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 6 cost path from unknown to gsm, via 1
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to gsm, via 3
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:1209 retrans_pkt: ** SIP timers: Rescheduling retransmission 3 to 2000 ms (t1 500 ms (Retrans id #47)) 
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 71f439ca643c96aa63da468e6d077ab4@24.123.23.170 Their Tag  Our tag: as19738910
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae6647666e@24.123.23.170 Their Tag 1c28ba8194fa9bb96cd0ff11cef75bd0 Our tag: as5b6350dc
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3168 sip_alloc: Allocating new SIP dialog for 000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165 - REGISTER (No RTP)
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:01:19]   == Registered translator 'lintoulaw' from format slin to ulaw, cost 1
[Mar 10 12:01:19] DEBUG[15131]: translate.c:278 rebuild_matrix: Resetting translation matrix
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from g723 to gsm, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from g723 to ulaw, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from g723 to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from gsm to g723, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from gsm to ulaw, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from gsm to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from gsm to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from gsm to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 2
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 2
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 2
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 2
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 2
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to ulaw, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from unknown to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from ulaw to g723, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from ulaw to gsm, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from ulaw to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from ulaw to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from ulaw to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from ulaw to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to gsm, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to ulaw, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from unknown to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to gsm, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to ulaw, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 14 cost path from unknown to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to gsm, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6
[Mar 10 12:01:19] DEBUG[15131]: translate.c:322 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = Found Their Call ID: 000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165 Their Tag  Our tag: as58fc5493
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:11239 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 11578, ours 11578)

[Mar 10 12:01:19]  [app_lookupblacklist.so][Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165 Their Tag  Our tag: as58fc5493
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 71f439ca643c96aa63da468e6d077ab4@24.123.23.170 Their Tag  Our tag: as19738910
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = Found Their Call ID: 55ae6647666e@24.123.23.170 Their Tag 1c28ba8194fa9bb96cd0ff11cef75bd0 Our tag: as5b6350dc
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:11239 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 24024, ours 24024)
[Mar 10 12:01:19]  => (Look up Caller*ID name/number from blacklist database)

[Mar 10 12:01:19]   == Registered application 'LookupBlacklist'

[Mar 10 12:01:19]  [app_getcpeid.so][Mar 10 12:01:19] DEBUG[15148]: chan_iax2.c:6670 socket_read: Packet arrived out of order (expecting 2, got 1) (frametype = 6, subclass = 13)
[Mar 10 12:01:19] DEBUG[15148]: chan_iax2.c:6677 socket_read: Acking anyway
[Mar 10 12:01:19] DEBUG[15148]: chan_iax2.c:6670 socket_read: Packet arrived out of order (expecting 2, got 1) (frametype = 6, subclass = 13)
[Mar 10 12:01:19] DEBUG[15148]: chan_iax2.c:6677 socket_read: Acking anyway
[Mar 10 12:01:19]  => (Get ADSI CPE ID)

[Mar 10 12:01:19]   == Registered application 'GetCPEID'

[Mar 10 12:01:19]  [format_wav.so][Mar 10 12:01:19] DEBUG[15136]: chan_iax2.c:9448 iax2_devicestate: Checking device state for device 606
[Mar 10 12:01:19] DEBUG[15136]: chan_iax2.c:9456 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 606? addr=-1509840704, defaddr=0 maxms=0, lastms=0
[Mar 10 12:01:19] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for IAX2/606 - state 1 (Not in use)
[Mar 10 12:01:19] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 526
[Mar 10 12:01:19] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/526 - state 1 (Not in use)
[Mar 10 12:01:19] DEBUG[15136]: chan_iax2.c:9448 iax2_devicestate: Checking device state for device 616
[Mar 10 12:01:19]  => (Microsoft WAV format (8000hz Signed Linear))
[Mar 10 12:01:19] DEBUG[15136]: chan_iax2.c:9456 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 616? addr=1159759178, defaddr=0 maxms=0, lastms=0

[Mar 10 12:01:19]   == Registered file format wav, extension(s) wav

[Mar 10 12:01:19]  [app_sms.so][Mar 10 12:01:19]  => (SMS/PSTN handler)

[Mar 10 12:01:19]   == Registered application 'SMS'

[Mar 10 12:01:19]  [app_authenticate.so][Mar 10 12:01:19]  => (Authentication Application)

[Mar 10 12:01:19]   == Registered application 'Authenticate'

[Mar 10 12:01:19] Asterisk Ready.
[Mar 10 12:01:19] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for IAX2/616 - state 1 (Not in use)
]1;Asterisk]2;Asterisk Console on 'unifiedpaging.messagenetsystems.com' (pid 15131)[Mar 10 12:01:19] DEBUG[15136]: chan_iax2.c:9448 iax2_devicestate: Checking device state for device 651
[Mar 10 12:01:19] DEBUG[15136]: chan_iax2.c:9456 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 651? addr=-1459509056, defaddr=0 maxms=0, lastms=0
[Mar 10 12:01:19] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for IAX2/651 - state 1 (Not in use)
[Mar 10 12:01:19] DEBUG[15161]: app_queue.c:500 changethread: Device 'IAX2/606' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
*CLI> [Mar 10 12:01:19] DEBUG[15163]: app_queue.c:500 changethread: Device 'IAX2/616' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:01:19] DEBUG[15162]: app_queue.c:500 changethread: Device 'SIP/526' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:01:19] DEBUG[15164]: app_queue.c:500 changethread: Device 'IAX2/651' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:01:19] DEBUG[15148]: db.c:200 ast_db_get: Unable to find key 'si-000364000738' in family 'iax/provisioning/cache'
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165 Their Tag  Our tag: as58fc5493
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 71f439ca643c96aa63da468e6d077ab4@24.123.23.170 Their Tag  Our tag: as19738910
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae6647666e@24.123.23.170 Their Tag 1c28ba8194fa9bb96cd0ff11cef75bd0 Our tag: as5b6350dc
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3168 sip_alloc: Allocating new SIP dialog for 1319586744@192.168.1.97 - REGISTER (No RTP)
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER
[Mar 10 12:01:19] DEBUG[15148]: iax2-provision.c:249 iax_provision_version: Unable to create provisioning packet for 'si-000364000738'
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 1319586744@192.168.1.97 Their Tag 2205145564 Our tag: as5f661c56
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = Found Their Call ID: 000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165 Their Tag  Our tag: as58fc5493
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:11239 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 11578, ours 11578)
[Mar 10 12:01:19] DEBUG[15148]: db.c:200 ast_db_get: Unable to find key 'si-000fd3000028' in family 'iax/provisioning/cache'
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 1319586744@192.168.1.97 Their Tag 2205145564 Our tag: as5f661c56
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165 Their Tag  Our tag: as58fc5493
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 71f439ca643c96aa63da468e6d077ab4@24.123.23.170 Their Tag  Our tag: as19738910
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = Found Their Call ID: 55ae6647666e@24.123.23.170 Their Tag 1c28ba8194fa9bb96cd0ff11cef75bd0 Our tag: as5b6350dc
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:11239 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 24024, ours 24024)
[Mar 10 12:01:19] DEBUG[15148]: iax2-provision.c:249 iax_provision_version: Unable to create provisioning packet for 'si-000fd3000028'
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 1319586744@192.168.1.97 Their Tag 2205145564 Our tag: as5f661c56
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165 Their Tag  Our tag: as58fc5493
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = Found Their Call ID: 71f439ca643c96aa63da468e6d077ab4@24.123.23.170 Their Tag  Our tag: as19738910
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:1412 __sip_ack: Stopping retransmission on '71f439ca643c96aa63da468e6d077ab4@24.123.23.170' of Request 102: Match Found
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:9843 handle_response_register: Registration successful
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:9845 handle_response_register: Cancelling timeout 46
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 1319586744@192.168.1.97 Their Tag 2205145564 Our tag: as5f661c56
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165 Their Tag  Our tag: as58fc5493
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 71f439ca643c96aa63da468e6d077ab4@24.123.23.170 Their Tag  Our tag: as19738910
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae6647666e@24.123.23.170 Their Tag 1c28ba8194fa9bb96cd0ff11cef75bd0 Our tag: as5b6350dc
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3168 sip_alloc: Allocating new SIP dialog for 55ae664767e0@24.123.23.170 - REGISTER (No RTP)
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae664767e0@24.123.23.170 Their Tag 7db4f3a1e6a327cfacdec467e70802d3 Our tag: as2d688225
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 1319586744@192.168.1.97 Their Tag 2205145564 Our tag: as5f661c56
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = Found Their Call ID: 000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165 Their Tag  Our tag: as58fc5493
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:11239 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 11578, ours 11578)
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae664767e0@24.123.23.170 Their Tag 7db4f3a1e6a327cfacdec467e70802d3 Our tag: as2d688225
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 1319586744@192.168.1.97 Their Tag 2205145564 Our tag: as5f661c56
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165 Their Tag  Our tag: as58fc5493
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 71f439ca643c96aa63da468e6d077ab4@24.123.23.170 Their Tag  Our tag: as19738910
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = Found Their Call ID: 55ae6647666e@24.123.23.170 Their Tag 1c28ba8194fa9bb96cd0ff11cef75bd0 Our tag: as5b6350dc
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:11239 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 24024, ours 24024)
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae664767e0@24.123.23.170 Their Tag 7db4f3a1e6a327cfacdec467e70802d3 Our tag: as2d688225
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 1319586744@192.168.1.97 Their Tag 2205145564 Our tag: as5f661c56
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165 Their Tag  Our tag: as58fc5493
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 71f439ca643c96aa63da468e6d077ab4@24.123.23.170 Their Tag  Our tag: as19738910
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae6647666e@24.123.23.170 Their Tag 1c28ba8194fa9bb96cd0ff11cef75bd0 Our tag: as5b6350dc
[Mar 10 12:01:19] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 526
[Mar 10 12:01:19] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/526 - state 1 (Not in use)
[Mar 10 12:01:19] DEBUG[15136]: chan_iax2.c:9448 iax2_devicestate: Checking device state for device 606
[Mar 10 12:01:19] DEBUG[15136]: chan_iax2.c:9456 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 606? addr=-1509840704, defaddr=0 maxms=0, lastms=0
[Mar 10 12:01:19] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for IAX2/606 - state 1 (Not in use)
[Mar 10 12:01:19] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 526
[Mar 10 12:01:19] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/526 - state 1 (Not in use)
[Mar 10 12:01:19] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 526
[Mar 10 12:01:19] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/526 - state 1 (Not in use)
[Mar 10 12:01:19] DEBUG[15165]: app_queue.c:500 changethread: Device 'SIP/526' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:01:19] DEBUG[15166]: app_queue.c:500 changethread: Device 'IAX2/606' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:01:19] DEBUG[15167]: app_queue.c:500 changethread: Device 'SIP/526' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:01:19] DEBUG[15168]: app_queue.c:500 changethread: Device 'SIP/526' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3168 sip_alloc: Allocating new SIP dialog for 55ae66469f7f@24.123.23.170 - REGISTER (No RTP)
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae66469f7f@24.123.23.170 Their Tag dba87193eb0d2fd82a96b38a6a348672 Our tag: as20acab1f
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae664767e0@24.123.23.170 Their Tag 7db4f3a1e6a327cfacdec467e70802d3 Our tag: as2d688225
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 1319586744@192.168.1.97 Their Tag 2205145564 Our tag: as5f661c56
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165 Their Tag  Our tag: as58fc5493
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 71f439ca643c96aa63da468e6d077ab4@24.123.23.170 Their Tag  Our tag: as19738910
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae6647666e@24.123.23.170 Their Tag 1c28ba8194fa9bb96cd0ff11cef75bd0 Our tag: as5b6350dc
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3168 sip_alloc: Allocating new SIP dialog for 55ae664696c0@24.123.23.170 - REGISTER (No RTP)
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae664696c0@24.123.23.170 Their Tag 0d5e9c7aa6ab28b144c66ddee0e32c88 Our tag: as133b683d
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae66469f7f@24.123.23.170 Their Tag dba87193eb0d2fd82a96b38a6a348672 Our tag: as20acab1f
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae664767e0@24.123.23.170 Their Tag 7db4f3a1e6a327cfacdec467e70802d3 Our tag: as2d688225
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 1319586744@192.168.1.97 Their Tag 2205145564 Our tag: as5f661c56
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165 Their Tag  Our tag: as58fc5493
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = Found Their Call ID: 71f439ca643c96aa63da468e6d077ab4@24.123.23.170 Their Tag  Our tag: as19738910
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:1412 __sip_ack: Stopping retransmission on '71f439ca643c96aa63da468e6d077ab4@24.123.23.170' of Request 102: Match Not Found
[Mar 10 12:01:19] WARNING[15149]: chan_sip.c:9835 handle_response_register: Got 200 OK on REGISTER that isn't a register
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae664696c0@24.123.23.170 Their Tag 0d5e9c7aa6ab28b144c66ddee0e32c88 Our tag: as133b683d
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae66469f7f@24.123.23.170 Their Tag dba87193eb0d2fd82a96b38a6a348672 Our tag: as20acab1f
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae664767e0@24.123.23.170 Their Tag 7db4f3a1e6a327cfacdec467e70802d3 Our tag: as2d688225
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 1319586744@192.168.1.97 Their Tag 2205145564 Our tag: as5f661c56
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165 Their Tag  Our tag: as58fc5493
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae6647666e@24.123.23.170 Their Tag 1c28ba8194fa9bb96cd0ff11cef75bd0 Our tag: as5b6350dc
[Mar 10 12:01:19] DEBUG[15136]: chan_iax2.c:9448 iax2_devicestate: Checking device state for device 616
[Mar 10 12:01:19] DEBUG[15136]: chan_iax2.c:9456 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 616? addr=1159759178, defaddr=0 maxms=0, lastms=0
[Mar 10 12:01:19] DEBUG[15148]: db.c:200 ast_db_get: Unable to find key 'si-000fd3000124' in family 'iax/provisioning/cache'
[Mar 10 12:01:19] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for IAX2/616 - state 1 (Not in use)
[Mar 10 12:01:19] DEBUG[15169]: app_queue.c:500 changethread: Device 'IAX2/616' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:01:19] DEBUG[15148]: iax2-provision.c:249 iax_provision_version: Unable to create provisioning packet for 'si-000fd3000124'
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae664696c0@24.123.23.170 Their Tag 0d5e9c7aa6ab28b144c66ddee0e32c88 Our tag: as133b683d
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae66469f7f@24.123.23.170 Their Tag dba87193eb0d2fd82a96b38a6a348672 Our tag: as20acab1f
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae664767e0@24.123.23.170 Their Tag 7db4f3a1e6a327cfacdec467e70802d3 Our tag: as2d688225
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 1319586744@192.168.1.97 Their Tag 2205145564 Our tag: as5f661c56
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = Found Their Call ID: 000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165 Their Tag  Our tag: as58fc5493
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:01:19]     -- Saved useragent "CSCO/7" for peer 511
[Mar 10 12:01:19] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 511
[Mar 10 12:01:19] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/511 - state 1 (Not in use)
[Mar 10 12:01:19] DEBUG[15170]: app_queue.c:500 changethread: Device 'SIP/511' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae664696c0@24.123.23.170 Their Tag 0d5e9c7aa6ab28b144c66ddee0e32c88 Our tag: as133b683d
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae66469f7f@24.123.23.170 Their Tag dba87193eb0d2fd82a96b38a6a348672 Our tag: as20acab1f
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae664767e0@24.123.23.170 Their Tag 7db4f3a1e6a327cfacdec467e70802d3 Our tag: as2d688225
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = Found Their Call ID: 1319586744@192.168.1.97 Their Tag 2205145564 Our tag: as5f661c56
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:11239 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 50, ours 50)
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae664696c0@24.123.23.170 Their Tag 0d5e9c7aa6ab28b144c66ddee0e32c88 Our tag: as133b683d
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae66469f7f@24.123.23.170 Their Tag dba87193eb0d2fd82a96b38a6a348672 Our tag: as20acab1f
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae664767e0@24.123.23.170 Their Tag 7db4f3a1e6a327cfacdec467e70802d3 Our tag: as2d688225
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = Found Their Call ID: 1319586744@192.168.1.97 Their Tag 2205145564 Our tag: as5f661c56
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:01:19]     -- Saved useragent "UTSTARCOM F1000/Device ID-0007ba26174b" for peer 551
[Mar 10 12:01:19] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 551
[Mar 10 12:01:19] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/551 - state 1 (Not in use)
[Mar 10 12:01:19] DEBUG[15171]: app_queue.c:500 changethread: Device 'SIP/551' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae664696c0@24.123.23.170 Their Tag 0d5e9c7aa6ab28b144c66ddee0e32c88 Our tag: as133b683d
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = Found Their Call ID: 55ae66469f7f@24.123.23.170 Their Tag dba87193eb0d2fd82a96b38a6a348672 Our tag: as20acab1f
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:01:19]     -- Saved useragent "Uniden SIP Phone p2 Ver BS4.77" for peer 528
[Mar 10 12:01:19] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 528
[Mar 10 12:01:19] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/528 - state 1 (Not in use)
[Mar 10 12:01:19] DEBUG[15172]: app_queue.c:500 changethread: Device 'SIP/528' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:3216 find_call: = Found Their Call ID: 55ae664696c0@24.123.23.170 Their Tag 0d5e9c7aa6ab28b144c66ddee0e32c88 Our tag: as133b683d
[Mar 10 12:01:19] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:01:19]     -- Saved useragent "Uniden SIP Phone p2 Ver BS4.77" for peer 529
[Mar 10 12:01:19] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 529
[Mar 10 12:01:19] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/529 - state 1 (Not in use)
[Mar 10 12:01:19] DEBUG[15173]: app_queue.c:500 changethread: Device 'SIP/529' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:01:21] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae664696c0@24.123.23.170 Their Tag 0d5e9c7aa6ab28b144c66ddee0e32c88 Our tag: as133b683d
[Mar 10 12:01:21] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae66469f7f@24.123.23.170 Their Tag dba87193eb0d2fd82a96b38a6a348672 Our tag: as20acab1f
[Mar 10 12:01:21] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae664767e0@24.123.23.170 Their Tag 7db4f3a1e6a327cfacdec467e70802d3 Our tag: as2d688225
[Mar 10 12:01:21] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 1319586744@192.168.1.97 Their Tag 2205145564 Our tag: as5f661c56
[Mar 10 12:01:21] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165 Their Tag  Our tag: as58fc5493
[Mar 10 12:01:21] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae6647666e@24.123.23.170 Their Tag 1c28ba8194fa9bb96cd0ff11cef75bd0 Our tag: as5b6350dc
[Mar 10 12:01:21] DEBUG[15149]: chan_sip.c:3168 sip_alloc: Allocating new SIP dialog for 000ff78d-ebb20002-23a7f663-5bd85286@192.168.1.165 - REGISTER (No RTP)
[Mar 10 12:01:21] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER
[Mar 10 12:01:21] DEBUG[15149]: chan_sip.c:3216 find_call: = Found Their Call ID: 000ff78d-ebb20002-23a7f663-5bd85286@192.168.1.165 Their Tag  Our tag: as2e2a3974
[Mar 10 12:01:21] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:01:21]     -- Saved useragent "CSCO/7" for peer 510
[Mar 10 12:01:21] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 510
[Mar 10 12:01:21] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/510 - state 1 (Not in use)
[Mar 10 12:01:21] DEBUG[15174]: app_queue.c:500 changethread: Device 'SIP/510' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.

*CLI> setd[Mar 10 12:01:29] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 000ff78d-ebb20002-23a7f663-5bd85286@192.168.1.165 Their Tag  Our tag: as2e2a3974
[Mar 10 12:01:29] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae664696c0@24.123.23.170 Their Tag 0d5e9c7aa6ab28b144c66ddee0e32c88 Our tag: as133b683d
[Mar 10 12:01:29] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae66469f7f@24.123.23.170 Their Tag dba87193eb0d2fd82a96b38a6a348672 Our tag: as20acab1f
[Mar 10 12:01:29] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae664767e0@24.123.23.170 Their Tag 7db4f3a1e6a327cfacdec467e70802d3 Our tag: as2d688225
[Mar 10 12:01:29] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 1319586744@192.168.1.97 Their Tag 2205145564 Our tag: as5f661c56
[Mar 10 12:01:29] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165 Their Tag  Our tag: as58fc5493
[Mar 10 12:01:29] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae6647666e@24.123.23.170 Their Tag 1c28ba8194fa9bb96cd0ff11cef75bd0 Our tag: as5b6350dc
[Mar 10 12:01:29] DEBUG[15149]: chan_sip.c:3168 sip_alloc: Allocating new SIP dialog for 55ae66476793@24.123.23.170 - REGISTER (No RTP)
[Mar 10 12:01:29] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER
e[Mar 10 12:01:29] DEBUG[15149]: chan_sip.c:3216 find_call: = Found Their Call ID: 55ae66476793@24.123.23.170 Their Tag 015060b4419e2a8745271117fcd8e780 Our tag: as02ac3530
[Mar 10 12:01:29] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:01:29]     -- Saved useragent "Uniden SIP Phone p2 Ver BS4.77" for peer 525
[Mar 10 12:01:29] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 525
[Mar 10 12:01:29] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/525 - state 1 (Not in use)
[Mar 10 12:01:29] DEBUG[15175]: app_queue.c:500 changethread: Device 'SIP/525' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
bug 4
No such command 'setdebug' (type 'help' for help)
*CLI> set d[Mar 10 12:01:33] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae66476793@24.123.23.170 Their Tag 015060b4419e2a8745271117fcd8e780 Our tag: as02ac3530
[Mar 10 12:01:33] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 000ff78d-ebb20002-23a7f663-5bd85286@192.168.1.165 Their Tag  Our tag: as2e2a3974
[Mar 10 12:01:33] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae664696c0@24.123.23.170 Their Tag 0d5e9c7aa6ab28b144c66ddee0e32c88 Our tag: as133b683d
[Mar 10 12:01:33] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae66469f7f@24.123.23.170 Their Tag dba87193eb0d2fd82a96b38a6a348672 Our tag: as20acab1f
[Mar 10 12:01:33] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae664767e0@24.123.23.170 Their Tag 7db4f3a1e6a327cfacdec467e70802d3 Our tag: as2d688225
[Mar 10 12:01:33] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 1319586744@192.168.1.97 Their Tag 2205145564 Our tag: as5f661c56
[Mar 10 12:01:33] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165 Their Tag  Our tag: as58fc5493
[Mar 10 12:01:33] DEBUG[15149]: chan_sip.c:3216 find_call: = No match Their Call ID: 55ae6647666e@24.123.23.170 Their Tag 1c28ba8194fa9bb96cd0ff11cef75bd0 Our tag: as5b6350dc
[Mar 10 12:01:33] DEBUG[15149]: chan_sip.c:3168 sip_alloc: Allocating new SIP dialog for 55ae6647678c@24.123.23.170 - REGISTER (No RTP)
[Mar 10 12:01:33] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER
[Mar 10 12:01:33] DEBUG[15149]: chan_sip.c:3216 find_call: = Found Their Call ID: 55ae6647678c@24.123.23.170 Their Tag 531437139972283a1f566640b7e3a4e9 Our tag: as14524c40
[Mar 10 12:01:33] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:01:33]     -- Saved useragent "Uniden SIP Phone p2 Ver BS4.77" for peer 524
[Mar 10 12:01:33] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 524
[Mar 10 12:01:33] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/524 - state 1 (Not in use)
[Mar 10 12:01:33] DEBUG[15176]: app_queue.c:500 changethread: Device 'SIP/524' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:01:34] DEBUG[15149]: chan_sip.c:1334 __sip_autodestruct: Auto destroying call '55ae664767e0@24.123.23.170'
[Mar 10 12:01:34] DEBUG[15149]: chan_sip.c:1334 __sip_autodestruct: Auto destroying call '55ae6647666e@24.123.23.170'
[Mar 10 12:01:34] DEBUG[15149]: chan_sip.c:1334 __sip_autodestruct: Auto destroying call '000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165'
[Mar 10 12:01:34] DEBUG[15149]: chan_sip.c:1334 __sip_autodestruct: Auto destroying call '1319586744@192.168.1.97'
[Mar 10 12:01:34] DEBUG[15149]: chan_sip.c:1334 __sip_autodestruct: Auto destroying call '55ae66469f7f@24.123.23.170'
[Mar 10 12:01:34] DEBUG[15149]: chan_sip.c:1334 __sip_autodestruct: Auto destroying call '55ae664696c0@24.123.23.170'
ebug 4
Core debug was 6 and is now 4
*CLI> [Mar 10 12:01:35] DEBUG[15149]: chan_sip.c:3168 sip_alloc: Allocating new SIP dialog for 55ae66469f11@24.123.23.170 - REGISTER (No RTP)
[Mar 10 12:01:35] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER
[Mar 10 12:01:35] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:01:35]     -- Saved useragent "Uniden SIP Phone p2 Ver BS4.77" for peer 522
[Mar 10 12:01:35] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 522
[Mar 10 12:01:35] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/522 - state 1 (Not in use)
[Mar 10 12:01:35] DEBUG[15177]: app_queue.c:500 changethread: Device 'SIP/522' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:01:36] DEBUG[15149]: chan_sip.c:1334 __sip_autodestruct: Auto destroying call '000ff78d-ebb20002-23a7f663-5bd85286@192.168.1.165'
[Mar 10 12:01:37] DEBUG[15149]: chan_sip.c:3168 sip_alloc: Allocating new SIP dialog for 000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165 - REGISTER (No RTP)
[Mar 10 12:01:37] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER
[Mar 10 12:01:37] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:01:37]     -- Saved useragent "CSCO/7" for peer 514
[Mar 10 12:01:37] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 514
[Mar 10 12:01:37] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/514 - state 1 (Not in use)
[Mar 10 12:01:37] DEBUG[15178]: app_queue.c:500 changethread: Device 'SIP/514' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:01:38] DEBUG[15149]: chan_sip.c:3168 sip_alloc: Allocating new SIP dialog for 000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165 - REGISTER (No RTP)
[Mar 10 12:01:38] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER
[Mar 10 12:01:38] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:01:38]     -- Saved useragent "CSCO/7" for peer 513
[Mar 10 12:01:38] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 513
[Mar 10 12:01:38] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/513 - state 1 (Not in use)
[Mar 10 12:01:38] DEBUG[15179]: app_queue.c:500 changethread: Device 'SIP/513' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:01:40] DEBUG[15149]: chan_sip.c:3168 sip_alloc: Allocating new SIP dialog for 55ae664767e0@24.123.23.170 - REGISTER (No RTP)
[Mar 10 12:01:40] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER
[Mar 10 12:01:40] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:01:40]     -- Saved useragent "Uniden SIP Phone p2 Ver BS4.77" for peer 523
[Mar 10 12:01:40] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 523
[Mar 10 12:01:40] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/523 - state 1 (Not in use)
[Mar 10 12:01:40] DEBUG[15180]: app_queue.c:500 changethread: Device 'SIP/523' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:01:41] DEBUG[15149]: chan_sip.c:3168 sip_alloc: Allocating new SIP dialog for 000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165 - REGISTER (No RTP)
[Mar 10 12:01:41] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER
[Mar 10 12:01:41] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:01:41]     -- Saved useragent "CSCO/7" for peer 512
[Mar 10 12:01:41] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 512
[Mar 10 12:01:41] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/512 - state 1 (Not in use)
[Mar 10 12:01:41] DEBUG[15181]: app_queue.c:500 changethread: Device 'SIP/512' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:01:42] DEBUG[15149]: chan_sip.c:3168 sip_alloc: Allocating new SIP dialog for 000ff78d-ebb20007-4ad25281-15520cf9@192.168.1.165 - REGISTER (No RTP)
[Mar 10 12:01:42] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER
[Mar 10 12:01:42] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:01:42]     -- Saved useragent "CSCO/7" for peer 515
[Mar 10 12:01:42] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 515
[Mar 10 12:01:42] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/515 - state 1 (Not in use)
[Mar 10 12:01:42] DEBUG[15182]: app_queue.c:500 changethread: Device 'SIP/515' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:01:43] DEBUG[15149]: chan_sip.c:3168 sip_alloc: Allocating new SIP dialog for 71f439ca643c96aa63da468e6d077ab4@24.123.23.170 - REGISTER (No RTP)
[Mar 10 12:01:43] DEBUG[15149]: chan_sip.c:5564 transmit_register: Scheduled a registration timeout for sip.broadvoice.com id  #102 

[Mar 10 12:01:43] REGISTER attempt 1 to 3173241052@sip.broadvoice.com
[Mar 10 12:01:43] DEBUG[15149]: chan_sip.c:1412 __sip_ack: Stopping retransmission on '71f439ca643c96aa63da468e6d077ab4@24.123.23.170' of Request 103: Match Found
[Mar 10 12:01:43] DEBUG[15149]: chan_sip.c:9843 handle_response_register: Registration successful
[Mar 10 12:01:43] DEBUG[15149]: chan_sip.c:9845 handle_response_register: Cancelling timeout 102
set ver[Mar 10 12:01:44] DEBUG[15149]: chan_sip.c:1334 __sip_autodestruct: Auto destroying call '55ae66476793@24.123.23.170'
bose 4
Verbosity was 5 and is now 4
*CLI> [Mar 10 12:01:48] DEBUG[15149]: chan_sip.c:1334 __sip_autodestruct: Auto destroying call '55ae6647678c@24.123.23.170'
sip debug[Mar 10 12:01:50] DEBUG[15149]: chan_sip.c:1334 __sip_autodestruct: Auto destroying call '55ae66469f11@24.123.23.170'

SIP Debugging enabled
*CLI> 
[Mar 10 12:01:51] 
<-- SIP read from 192.168.1.86:5060: 


[Mar 10 12:01:51] --- (0 headers 0 lines) Nat keepalive ---
[Mar 10 12:01:52] DEBUG[15149]: chan_sip.c:1334 __sip_autodestruct: Auto destroying call '000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165'

[Mar 10 12:01:52] Destroying call '000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165'
[Mar 10 12:01:52] DEBUG[15136]: chan_iax2.c:9448 iax2_devicestate: Checking device state for device 616
[Mar 10 12:01:52] DEBUG[15136]: chan_iax2.c:9456 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 616? addr=1159759178, defaddr=0 maxms=0, lastms=0
[Mar 10 12:01:52] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for IAX2/616 - state 1 (Not in use)
[Mar 10 12:01:52] DEBUG[15186]: app_queue.c:500 changethread: Device 'IAX2/616' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:01:52] DEBUG[15136]: chan_iax2.c:9448 iax2_devicestate: Checking device state for device 616
[Mar 10 12:01:52] DEBUG[15136]: chan_iax2.c:9456 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 616? addr=1159759178, defaddr=0 maxms=0, lastms=0
[Mar 10 12:01:52] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for IAX2/616 - state 1 (Not in use)
[Mar 10 12:01:52] DEBUG[15187]: app_queue.c:500 changethread: Device 'IAX2/616' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:01:53] DEBUG[15149]: chan_sip.c:1334 __sip_autodestruct: Auto destroying call '000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165'

[Mar 10 12:01:53] Destroying call '000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165'
[Mar 10 12:01:53] DEBUG[15136]: chan_iax2.c:9448 iax2_devicestate: Checking device state for device 606
[Mar 10 12:01:53] DEBUG[15136]: chan_iax2.c:9456 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 606? addr=-1509840704, defaddr=0 maxms=0, lastms=0
[Mar 10 12:01:53] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for IAX2/606 - state 1 (Not in use)
[Mar 10 12:01:53] DEBUG[15188]: app_queue.c:500 changethread: Device 'IAX2/606' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:01:53] DEBUG[15136]: chan_iax2.c:9448 iax2_devicestate: Checking device state for device 606
[Mar 10 12:01:53] DEBUG[15136]: chan_iax2.c:9456 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 606? addr=-1509840704, defaddr=0 maxms=0, lastms=0
[Mar 10 12:01:53] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for IAX2/606 - state 1 (Not in use)
[Mar 10 12:01:53] DEBUG[15189]: app_queue.c:500 changethread: Device 'IAX2/606' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.

[Mar 10 12:01:54] 
<-- SIP read from 192.168.1.97:5060: 


[Mar 10 12:01:54] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0:  (0)

[Mar 10 12:01:54] --- (0 headers 0 lines) Nat keepalive ---
[Mar 10 12:01:55] DEBUG[15149]: chan_sip.c:1334 __sip_autodestruct: Auto destroying call '55ae664767e0@24.123.23.170'

[Mar 10 12:01:55] Destroying call '55ae664767e0@24.123.23.170'
[Mar 10 12:01:56] DEBUG[15149]: chan_sip.c:1334 __sip_autodestruct: Auto destroying call '000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165'

[Mar 10 12:01:56] Destroying call '000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165'
[Mar 10 12:01:57] DEBUG[15190]: manager.c:1249 process_message: Manager received command 'Login'

[Mar 10 12:01:57]   == Parsing '/etc/asterisk/manager.conf': [Mar 10 12:01:57] DEBUG[15190]: config.c:595 config_text_file_load: Parsing /etc/asterisk/manager.conf
[Mar 10 12:01:57] Found
[Mar 10 12:01:57] DEBUG[15190]: acl.c:199 ast_append_ha: 127.0.0.1/255.255.255.0/255.255.255.0 appended to acl for peer
[Mar 10 12:01:57] DEBUG[15190]: acl.c:211 ast_apply_ha: ##### Testing 127.0.0.1 with 127.0.0.0

[Mar 10 12:01:57]   == Manager 'MessageNet' logged on from 127.0.0.1
[Mar 10 12:01:57] DEBUG[15149]: chan_sip.c:1334 __sip_autodestruct: Auto destroying call '000ff78d-ebb20007-4ad25281-15520cf9@192.168.1.165'

[Mar 10 12:01:57] Destroying call '000ff78d-ebb20007-4ad25281-15520cf9@192.168.1.165'
[Mar 10 12:01:57] DEBUG[15190]: manager.c:1249 process_message: Manager received command 'Command'
[Mar 10 12:01:57] DEBUG[15190]: manager.c:1249 process_message: Manager received command 'Command'
[Mar 10 12:01:57] DEBUG[15190]: manager.c:1249 process_message: Manager received command 'Logoff'

[Mar 10 12:01:57]   == Manager 'MessageNet' logged off from 127.0.0.1
[Mar 10 12:02:00] DEBUG[15136]: chan_iax2.c:9448 iax2_devicestate: Checking device state for device 614
[Mar 10 12:02:00] DEBUG[15136]: chan_iax2.c:9456 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 614? addr=-1493063488, defaddr=0 maxms=0, lastms=0
[Mar 10 12:02:00] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for IAX2/614 - state 1 (Not in use)
[Mar 10 12:02:00] DEBUG[15191]: app_queue.c:500 changethread: Device 'IAX2/614' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:02:00] DEBUG[15136]: chan_iax2.c:9448 iax2_devicestate: Checking device state for device 614
[Mar 10 12:02:00] DEBUG[15136]: chan_iax2.c:9456 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 614? addr=-1493063488, defaddr=0 maxms=0, lastms=0
[Mar 10 12:02:00] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for IAX2/614 - state 1 (Not in use)
[Mar 10 12:02:00] DEBUG[15192]: app_queue.c:500 changethread: Device 'IAX2/614' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.

[Mar 10 12:02:01] 
<-- SIP read from 68.58.36.157:5060: 


[Mar 10 12:02:01] --- (0 headers 0 lines) Nat keepalive ---

[Mar 10 12:02:01] 
<-- SIP read from 192.168.1.86:5062: 


[Mar 10 12:02:01] --- (0 headers 0 lines) Nat keepalive ---

[Mar 10 12:02:01] 
<-- SIP read from 192.168.1.83:5060: 
INVITE sip:523@24.123.23.170:5060 SIP/2.0
Max-Forwards: 70
Content-Length: 292
Via: SIP/2.0/UDP 192.168.1.83:5060;branch=z9hG4bK25b777a46
Call-ID: 20a99672af308312d36264481f6b7f5a@192.168.1.83
From: 593 <sip:593@24.123.23.170:5060>;tag=b3eeb4c9225baf5
To: 523 <sip:523@24.123.23.170:5060>
CSeq: 75546810 INVITE
Supported: timer
Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE
Content-Type: application/sdp
Contact: 593 <sip:593@192.168.1.83:5060>
Supported: replaces
User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4

v=0
o=MxSIP 0 1158539950 IN IP4 192.168.1.83
s=SIP Call
c=IN IP4 192.168.1.83
t=0 0
m=audio 51000 RTP/AVP 0 18 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=silenceSupp:on - - - -

[Mar 10 12:02:01] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: INVITE sip:523@24.123.23.170:5060 SIP/2.0 (41)
[Mar 10 12:02:01] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Max-Forwards: 70 (16)
[Mar 10 12:02:01] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: Content-Length: 292 (19)
[Mar 10 12:02:01] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: Via: SIP/2.0/UDP 192.168.1.83:5060;branch=z9hG4bK25b777a46 (58)
[Mar 10 12:02:01] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Call-ID: 20a99672af308312d36264481f6b7f5a@192.168.1.83 (54)
[Mar 10 12:02:01] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: From: 593 <sip:593@24.123.23.170:5060>;tag=b3eeb4c9225baf5 (58)
[Mar 10 12:02:01] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: To: 523 <sip:523@24.123.23.170:5060> (36)
[Mar 10 12:02:01] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: CSeq: 75546810 INVITE (21)
[Mar 10 12:02:01] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Supported: timer (16)
[Mar 10 12:02:01] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE (55)
[Mar 10 12:02:01] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Content-Type: application/sdp (29)
[Mar 10 12:02:01] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11: Contact: 593 <sip:593@192.168.1.83:5060> (40)
[Mar 10 12:02:01] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 12: Supported: replaces (19)
[Mar 10 12:02:01] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 13: User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 (42)
[Mar 10 12:02:01] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 14:  (0)
[Mar 10 12:02:01] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: v=0 (3)
[Mar 10 12:02:01] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: o=MxSIP 0 1158539950 IN IP4 192.168.1.83 (40)
[Mar 10 12:02:01] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: s=SIP Call (10)
[Mar 10 12:02:01] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: c=IN IP4 192.168.1.83 (21)
[Mar 10 12:02:01] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: t=0 0 (5)
[Mar 10 12:02:01] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: m=audio 51000 RTP/AVP 0 18 8 101 (32)
[Mar 10 12:02:01] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:0 PCMU/8000 (20)
[Mar 10 12:02:01] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:18 G729/8000 (21)
[Mar 10 12:02:01] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:8 PCMA/8000 (20)
[Mar 10 12:02:01] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33)
[Mar 10 12:02:01] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=fmtp:101 0-15 (15)
[Mar 10 12:02:01] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=ptime:20 (10)
[Mar 10 12:02:01] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=sendrecv (10)
[Mar 10 12:02:01] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=silenceSupp:on - - - - (24)

[Mar 10 12:02:01] --- (14 headers 14 lines) ---
[Mar 10 12:02:01] DEBUG[15149]: chan_sip.c:3168 sip_alloc: Allocating new SIP dialog for 20a99672af308312d36264481f6b7f5a@192.168.1.83 - INVITE (With RTP)
[Mar 10 12:02:01] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received INVITE (5) - Command in SIP INVITE
[Mar 10 12:02:01] DEBUG[15149]: chan_sip.c:1015 parse_sip_options: Begin: parsing SIP "Supported: timer"
[Mar 10 12:02:01] DEBUG[15149]: chan_sip.c:1027 parse_sip_options: Found SIP option: -timer-
[Mar 10 12:02:01] DEBUG[15149]: chan_sip.c:1033 parse_sip_options: Matched SIP option: timer
[Mar 10 12:02:01] DEBUG[15149]: chan_sip.c:1044 parse_sip_options: * SIP extension value: 4 for call 20a99672af308312d36264481f6b7f5a@192.168.1.83

[Mar 10 12:02:01] Using INVITE request as basis request - 20a99672af308312d36264481f6b7f5a@192.168.1.83

[Mar 10 12:02:01] Sending to 192.168.1.83 : 5060 (non-NAT)
[Mar 10 12:02:01] DEBUG[15149]: chan_sip.c:7215 check_user_full: Setting NAT on RTP to 0
[Mar 10 12:02:01] DEBUG[15149]: chan_sip.c:7219 check_user_full: Setting NAT on VRTP to 0

[Mar 10 12:02:01] Reliably Transmitting (no NAT) to 192.168.1.83:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.83:5060;branch=z9hG4bK25b777a46;received=192.168.1.83
From: 593 <sip:593@24.123.23.170:5060>;tag=b3eeb4c9225baf5
To: 523 <sip:523@24.123.23.170:5060>;tag=as5ca0ef44
Call-ID: 20a99672af308312d36264481f6b7f5a@192.168.1.83
CSeq: 75546810 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4de95353"
Content-Length: 0


---
[Mar 10 12:02:01] DEBUG[15149]: chan_sip.c:1304 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id  #106

[Mar 10 12:02:01] Scheduling destruction of call '20a99672af308312d36264481f6b7f5a@192.168.1.83' in 15000 ms

[Mar 10 12:02:01] Found user '593'
[Mar 10 12:02:01] DEBUG[15136]: chan_iax2.c:9448 iax2_devicestate: Checking device state for device 651
[Mar 10 12:02:01] DEBUG[15136]: chan_iax2.c:9456 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 651? addr=-1459509056, defaddr=0 maxms=0, lastms=0
[Mar 10 12:02:01] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for IAX2/651 - state 1 (Not in use)
[Mar 10 12:02:01] DEBUG[15193]: app_queue.c:500 changethread: Device 'IAX2/651' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:02:01] DEBUG[15136]: chan_iax2.c:9448 iax2_devicestate: Checking device state for device 651
[Mar 10 12:02:01] DEBUG[15136]: chan_iax2.c:9456 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 651? addr=-1459509056, defaddr=0 maxms=0, lastms=0
[Mar 10 12:02:01] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for IAX2/651 - state 1 (Not in use)
[Mar 10 12:02:01] DEBUG[15194]: app_queue.c:500 changethread: Device 'IAX2/651' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.

[Mar 10 12:02:02] 
<-- SIP read from 192.168.1.83:5060: 
ACK sip:523@24.123.23.170:5060 SIP/2.0
Max-Forwards: 70
Content-Length: 0
Via: SIP/2.0/UDP 192.168.1.83:5060;branch=z9hG4bK25b777a46
Call-ID: 20a99672af308312d36264481f6b7f5a@192.168.1.83
From: 593 <sip:593@24.123.23.170:5060>;tag=b3eeb4c9225baf5
To: 523 <sip:523@24.123.23.170:5060>;tag=as5ca0ef44
CSeq: 75546810 ACK
User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4


[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: ACK sip:523@24.123.23.170:5060 SIP/2.0 (38)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Max-Forwards: 70 (16)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: Content-Length: 0 (17)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: Via: SIP/2.0/UDP 192.168.1.83:5060;branch=z9hG4bK25b777a46 (58)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Call-ID: 20a99672af308312d36264481f6b7f5a@192.168.1.83 (54)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: From: 593 <sip:593@24.123.23.170:5060>;tag=b3eeb4c9225baf5 (58)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: To: 523 <sip:523@24.123.23.170:5060>;tag=as5ca0ef44 (51)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: CSeq: 75546810 ACK (18)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 (42)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9:  (0)

[Mar 10 12:02:02] --- (9 headers 0 lines) ---
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received ACK (6) - Command in SIP ACK
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:1401 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #106
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:1412 __sip_ack: Stopping retransmission on '20a99672af308312d36264481f6b7f5a@192.168.1.83' of Response 75546810: Match Found

[Mar 10 12:02:02] 
<-- SIP read from 192.168.1.83:5060: 
INVITE sip:523@24.123.23.170:5060 SIP/2.0
Max-Forwards: 70
Content-Length: 292
Via: SIP/2.0/UDP 192.168.1.83:5060;branch=z9hG4bKe316288f6
Call-ID: 20a99672af308312d36264481f6b7f5a@192.168.1.83
From: 593 <sip:593@24.123.23.170:5060>;tag=b3eeb4c9225baf5
To: 523 <sip:523@24.123.23.170:5060>
CSeq: 75546811 INVITE
Supported: timer
Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE
Content-Type: application/sdp
Supported: replaces
Proxy-Authorization:Digest response="f2ad6f8c0955a18b6c8fd9a6bf574bdf",username="593",realm="asterisk",nonce="4de95353",algorithm=MD5,uri="sip:523@24.123.23.170:5060"
Contact: 593 <sip:593@192.168.1.83:5060>
User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4

v=0
o=MxSIP 0 1158539950 IN IP4 192.168.1.83
s=SIP Call
c=IN IP4 192.168.1.83
t=0 0
m=audio 51000 RTP/AVP 0 18 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=silenceSupp:on - - - -

[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: INVITE sip:523@24.123.23.170:5060 SIP/2.0 (41)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Max-Forwards: 70 (16)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: Content-Length: 292 (19)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: Via: SIP/2.0/UDP 192.168.1.83:5060;branch=z9hG4bKe316288f6 (58)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Call-ID: 20a99672af308312d36264481f6b7f5a@192.168.1.83 (54)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: From: 593 <sip:593@24.123.23.170:5060>;tag=b3eeb4c9225baf5 (58)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: To: 523 <sip:523@24.123.23.170:5060> (36)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: CSeq: 75546811 INVITE (21)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Supported: timer (16)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE (55)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Content-Type: application/sdp (29)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11: Supported: replaces (19)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 12: Proxy-Authorization:Digest response="f2ad6f8c0955a18b6c8fd9a6bf574bdf",username="593",realm="asterisk",nonce="4de95353",algorithm=MD5,uri="sip:523@24.123.23.170:5060" (166)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 13: Contact: 593 <sip:593@192.168.1.83:5060> (40)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 14: User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 (42)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 15:  (0)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: v=0 (3)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: o=MxSIP 0 1158539950 IN IP4 192.168.1.83 (40)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: s=SIP Call (10)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: c=IN IP4 192.168.1.83 (21)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: t=0 0 (5)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: m=audio 51000 RTP/AVP 0 18 8 101 (32)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:0 PCMU/8000 (20)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:18 G729/8000 (21)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:8 PCMA/8000 (20)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=fmtp:101 0-15 (15)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=ptime:20 (10)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=sendrecv (10)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=silenceSupp:on - - - - (24)

[Mar 10 12:02:02] --- (15 headers 14 lines) ---
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received INVITE (5) - Command in SIP INVITE

[Mar 10 12:02:02] Using INVITE request as basis request - 20a99672af308312d36264481f6b7f5a@192.168.1.83

[Mar 10 12:02:02] Sending to 192.168.1.83 : 5060 (non-NAT)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:7215 check_user_full: Setting NAT on RTP to 0
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:7219 check_user_full: Setting NAT on VRTP to 0

[Mar 10 12:02:02] Found user '593'

[Mar 10 12:02:02] Found RTP audio format 0

[Mar 10 12:02:02] Found RTP audio format 18

[Mar 10 12:02:02] Found RTP audio format 8

[Mar 10 12:02:02] Found RTP audio format 101

[Mar 10 12:02:02] Peer audio RTP is at port 192.168.1.83:51000
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3631 process_sdp: Peer audio RTP is at port 192.168.1.83:51000

[Mar 10 12:02:02] Peer video RTP is at port 192.168.1.83:65535
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3653 process_sdp: Peer video RTP is at port 192.168.1.83:65535

[Mar 10 12:02:02] Found description format PCMU

[Mar 10 12:02:02] Found description format G729

[Mar 10 12:02:02] Found description format PCMA

[Mar 10 12:02:02] Found description format telephone-event

[Mar 10 12:02:02] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)

[Mar 10 12:02:02] Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:10571 handle_request_invite: Checking SIP call limits for device 593
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:2217 update_call_counter: Updating call counter for incoming call

[Mar 10 12:02:02] Looking for 523 in smvoice-sip (domain 24.123.23.170)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:6194 build_route: build_route: Contact hop: 593 <sip:593@192.168.1.83:5060>

[Mar 10 12:02:02] list_route: hop: <sip:593@192.168.1.83:5060>

[Mar 10 12:02:02] Transmitting (no NAT) to 192.168.1.83:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.83:5060;branch=z9hG4bKe316288f6;received=192.168.1.83
From: 593 <sip:593@24.123.23.170:5060>;tag=b3eeb4c9225baf5
To: 523 <sip:523@24.123.23.170:5060>
Call-ID: 20a99672af308312d36264481f6b7f5a@192.168.1.83
CSeq: 75546811 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:523@24.123.23.170>
Content-Length: 0


---
[Mar 10 12:02:02] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 593
[Mar 10 12:02:02] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/593 - state 2 (In use)
[Mar 10 12:02:02] DEBUG[15196]: app_queue.c:500 changethread: Device 'SIP/593' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[Mar 10 12:02:02] DEBUG[15195]: pbx.c:1677 pbx_extension_helper: Launching 'NoOp'

[Mar 10 12:02:02]     -- Executing NoOp("SIP/593-084f44b8", "5xx") in new stack
[Mar 10 12:02:02] DEBUG[15195]: pbx.c:1677 pbx_extension_helper: Launching 'Set'

[Mar 10 12:02:02]     -- Executing Set("SIP/593-084f44b8", "SMVOICE_CONTEXT_EXTEN=523") in new stack
[Mar 10 12:02:02] DEBUG[15195]: pbx.c:1677 pbx_extension_helper: Launching 'AGI'

[Mar 10 12:02:02]     -- Executing AGI("SIP/593-084f44b8", "smvoice|-company|demonstration|-digium_asterisk|-asterisk_callat_forwarding|523") in new stack

[Mar 10 12:02:02]     -- Launched AGI Script /var/lib/asterisk/agi-bin/smvoice

[Mar 10 12:02:02]     -- AGI Script smvoice completed, returning 0
[Mar 10 12:02:02] DEBUG[15195]: pbx.c:1589 pbx_substitute_variables_helper_full: Expression result is '0'
[Mar 10 12:02:02] DEBUG[15195]: pbx.c:1677 pbx_extension_helper: Launching 'GotoIf'

[Mar 10 12:02:02]     -- Executing GotoIf("SIP/593-084f44b8", "0?INVALID|1") in new stack
[Mar 10 12:02:02] DEBUG[15195]: pbx.c:6177 pbx_builtin_gotoif: Not taking any branch
[Mar 10 12:02:02] DEBUG[15195]: pbx.c:1589 pbx_substitute_variables_helper_full: Expression result is '0'
[Mar 10 12:02:02] DEBUG[15195]: pbx.c:1677 pbx_extension_helper: Launching 'GotoIf'

[Mar 10 12:02:02]     -- Executing GotoIf("SIP/593-084f44b8", "0?_5XX-NOANSWER|1") in new stack
[Mar 10 12:02:02] DEBUG[15195]: pbx.c:6177 pbx_builtin_gotoif: Not taking any branch
[Mar 10 12:02:02] DEBUG[15195]: pbx.c:1677 pbx_extension_helper: Launching 'Dial'

[Mar 10 12:02:02]     -- Executing Dial("SIP/593-084f44b8", "SIP/523|20|") in new stack
[Mar 10 12:02:02] DEBUG[15195]: chan_sip.c:3168 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP)
[Mar 10 12:02:02] DEBUG[15195]: chan_sip.c:1885 create_addr_from_peer: Setting NAT on RTP to 0
[Mar 10 12:02:02] DEBUG[15195]: chan_sip.c:1889 create_addr_from_peer: Setting NAT on VRTP to 0
[Mar 10 12:02:02] DEBUG[15195]: channel.c:2902 ast_channel_inherit_variables: Not copying variable STACK-smvoice-sip-523-6.
[Mar 10 12:02:02] DEBUG[15195]: channel.c:2902 ast_channel_inherit_variables: Not copying variable STACK-smvoice-sip-523-5.
[Mar 10 12:02:02] DEBUG[15195]: channel.c:2902 ast_channel_inherit_variables: Not copying variable STACK-smvoice-sip-523-4.
[Mar 10 12:02:02] DEBUG[15195]: channel.c:2902 ast_channel_inherit_variables: Not copying variable SMVOICE_WORK_EXTENSION.
[Mar 10 12:02:02] DEBUG[15195]: channel.c:2902 ast_channel_inherit_variables: Not copying variable SMVOICE_EXTENSION.
[Mar 10 12:02:02] DEBUG[15195]: channel.c:2902 ast_channel_inherit_variables: Not copying variable SMVOICE_PIN.
[Mar 10 12:02:02] DEBUG[15195]: channel.c:2902 ast_channel_inherit_variables: Not copying variable SMVOICE_CALLAT.
[Mar 10 12:02:02] DEBUG[15195]: channel.c:2902 ast_channel_inherit_variables: Not copying variable STACK-smvoice-sip-523-3.
[Mar 10 12:02:02] DEBUG[15195]: channel.c:2902 ast_channel_inherit_variables: Not copying variable SMVOICE_CONTEXT_EXTEN.
[Mar 10 12:02:02] DEBUG[15195]: channel.c:2902 ast_channel_inherit_variables: Not copying variable STACK-smvoice-sip-523-2.
[Mar 10 12:02:02] DEBUG[15195]: channel.c:2902 ast_channel_inherit_variables: Not copying variable STACK-smvoice-sip-523-1.
[Mar 10 12:02:02] DEBUG[15195]: channel.c:2902 ast_channel_inherit_variables: Not copying variable SIPCALLID.
[Mar 10 12:02:02] DEBUG[15195]: channel.c:2902 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT.
[Mar 10 12:02:02] DEBUG[15195]: channel.c:2902 ast_channel_inherit_variables: Not copying variable SIPDOMAIN.
[Mar 10 12:02:02] DEBUG[15195]: channel.c:2902 ast_channel_inherit_variables: Not copying variable SIPURI.
[Mar 10 12:02:02] DEBUG[15195]: chan_sip.c:2079 sip_call: Outgoing Call for 523
[Mar 10 12:02:02] DEBUG[15195]: chan_sip.c:2217 update_call_counter: Updating call counter for outgoing call

[Mar 10 12:02:02] We're at 24.123.23.170 port 17986

[Mar 10 12:02:02] Video is at 24.123.23.170 port 12600

[Mar 10 12:02:02] Adding codec 0x4 (ulaw) to SDP

[Mar 10 12:02:02] Adding codec 0x8 (alaw) to SDP

[Mar 10 12:02:02] Adding codec 0x2 (gsm) to SDP

[Mar 10 12:02:02] Adding non-codec 0x1 (telephone-event) to SDP
[Mar 10 12:02:02] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 0: INVITE sip:523@192.168.1.85:5060 SIP/2.0 (40)
[Mar 10 12:02:02] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK7672cbff (58)
[Mar 10 12:02:02] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 2: From: "593 593" <sip:593@24.123.23.170>;tag=as0da2dd59 (54)
[Mar 10 12:02:02] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 3: To: <sip:523@192.168.1.85:5060> (31)
[Mar 10 12:02:02] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 4: Contact: <sip:593@24.123.23.170> (32)
[Mar 10 12:02:02] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 5: Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170 (55)
[Mar 10 12:02:02] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 6: CSeq: 102 INVITE (16)
[Mar 10 12:02:02] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 7: User-Agent: Asterisk PBX (24)
[Mar 10 12:02:02] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 8: Max-Forwards: 70 (16)
[Mar 10 12:02:02] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 9: Date: Sat, 10 Mar 2007 17:02:02 GMT (35)
[Mar 10 12:02:02] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66)
[Mar 10 12:02:02] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 11: Content-Type: application/sdp (29)
[Mar 10 12:02:02] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 12: Content-Length: 265 (19)
[Mar 10 12:02:02] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 13:  (0)
[Mar 10 12:02:02] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: v=0 (3)
[Mar 10 12:02:02] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: o=root 15131 15131 IN IP4 24.123.23.170 (39)
[Mar 10 12:02:02] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: s=session (9)
[Mar 10 12:02:02] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: c=IN IP4 24.123.23.170 (22)
[Mar 10 12:02:02] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: t=0 0 (5)
[Mar 10 12:02:02] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: m=audio 17986 RTP/AVP 0 8 3 101 (31)
[Mar 10 12:02:02] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: a=rtpmap:0 PCMU/8000 (20)
[Mar 10 12:02:02] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: a=rtpmap:8 PCMA/8000 (20)
[Mar 10 12:02:02] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: a=rtpmap:3 GSM/8000 (19)
[Mar 10 12:02:02] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33)
[Mar 10 12:02:02] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: a=fmtp:101 0-16 (15)
[Mar 10 12:02:02] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: a=silenceSupp:off - - - - (25)

[Mar 10 12:02:02] 13 headers, 12 lines

[Mar 10 12:02:02] Reliably Transmitting (no NAT) to 192.168.1.85:5060:
INVITE sip:523@192.168.1.85:5060 SIP/2.0
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK7672cbff
From: "593 593" <sip:593@24.123.23.170>;tag=as0da2dd59
To: <sip:523@192.168.1.85:5060>
Contact: <sip:593@24.123.23.170>
Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 10 Mar 2007 17:02:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 15131 15131 IN IP4 24.123.23.170
s=session
c=IN IP4 24.123.23.170
t=0 0
m=audio 17986 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 - - - -

---
[Mar 10 12:02:02] DEBUG[15195]: chan_sip.c:1304 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id  #108

[Mar 10 12:02:02]     -- Called 523

[Mar 10 12:02:02] 
<-- SIP read from 192.168.1.85:5060: 
SIP/2.0 100 Trying
To: <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK7672cbff
From: "593 593" <sip:593@24.123.23.170>;tag=as0da2dd59
Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170
CSeq: 102 INVITE
Content-Length: 0
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK


[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: SIP/2.0 100 Trying (18)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: To: <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f (68)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK7672cbff (58)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: From: "593 593" <sip:593@24.123.23.170>;tag=as0da2dd59 (54)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170 (55)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: CSeq: 102 INVITE (16)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: Content-Length: 0 (17)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Supported: sip-cc, sip-cc-01, replaces, timer (45)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9:  (0)

[Mar 10 12:02:02] --- (9 headers 0 lines) ---
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:1456 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #108 - INVITE (got response)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:1465 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '5d28f46f54fd450475a9f02419f99674@24.123.23.170' Request 102: Found
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:9642 handle_response_invite: SIP response 100 to standard invite

[Mar 10 12:02:02] 
<-- SIP read from 192.168.1.85:5060: 
SIP/2.0 180 Ringing
To: <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f
Contact: <sip:523@192.168.1.85:5060>
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK7672cbff
From: "593 593" <sip:593@24.123.23.170>;tag=as0da2dd59
Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170
CSeq: 102 INVITE
Content-Length: 0
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK


[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: SIP/2.0 180 Ringing (19)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: To: <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f (68)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: Contact: <sip:523@192.168.1.85:5060> (36)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK7672cbff (58)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: From: "593 593" <sip:593@24.123.23.170>;tag=as0da2dd59 (54)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170 (55)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: CSeq: 102 INVITE (16)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Content-Length: 0 (17)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Supported: sip-cc, sip-cc-01, replaces, timer (45)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62)
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10:  (0)

[Mar 10 12:02:02] --- (10 headers 0 lines) ---
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:1465 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '5d28f46f54fd450475a9f02419f99674@24.123.23.170' Request 102: Found
[Mar 10 12:02:02] DEBUG[15149]: chan_sip.c:9642 handle_response_invite: SIP response 180 to standard invite

[Mar 10 12:02:02]     -- SIP/523-0850b038 is ringing

[Mar 10 12:02:02] Transmitting (no NAT) to 192.168.1.83:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.83:5060;branch=z9hG4bKe316288f6;received=192.168.1.83
From: 593 <sip:593@24.123.23.170:5060>;tag=b3eeb4c9225baf5
To: 523 <sip:523@24.123.23.170:5060>;tag=as38239579
Call-ID: 20a99672af308312d36264481f6b7f5a@192.168.1.83
CSeq: 75546811 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:523@24.123.23.170>
Content-Length: 0


---
[Mar 10 12:02:02] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 523
[Mar 10 12:02:02] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/523 - state 6 (Ringing)
[Mar 10 12:02:02] DEBUG[15198]: app_queue.c:500 changethread: Device 'SIP/523' changed to state '6' (Ringing) but we don't care because they're not a member of any queue.

[Mar 10 12:02:05] 
<-- SIP read from 192.168.1.85:5060: 
SIP/2.0 200 OK
To: <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f
Contact: <sip:523@192.168.1.85:5060>
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK7672cbff
From: "593 593" <sip:593@24.123.23.170>;tag=as0da2dd59
Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170
CSeq: 102 INVITE
Content-Type: application/sdp
Content-Length: 223
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK

v=0
o=- 15131 248055 IN IP4 192.168.1.85
s=session
c=IN IP4 192.168.1.85
t=0 0
m=audio 30012 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,16,17,18
a=silenceSupp:off - - - -

[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: SIP/2.0 200 OK (14)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: To: <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f (68)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: Contact: <sip:523@192.168.1.85:5060> (36)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK7672cbff (58)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: From: "593 593" <sip:593@24.123.23.170>;tag=as0da2dd59 (54)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170 (55)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: CSeq: 102 INVITE (16)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Content-Type: application/sdp (29)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Content-Length: 223 (19)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Supported: sip-cc, sip-cc-01, replaces, timer (45)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11:  (0)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: v=0 (3)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: o=- 15131 248055 IN IP4 192.168.1.85 (36)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: s=session (9)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: c=IN IP4 192.168.1.85 (21)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: t=0 0 (5)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: m=audio 30012 RTP/AVP 0 101 (27)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:0 PCMU/8000 (20)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=fmtp:101 0-15,16,17,18 (24)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=silenceSupp:off - - - - (25)

[Mar 10 12:02:05] --- (11 headers 10 lines) ---
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:1390 __sip_ack: Acked pending invite 102
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:1412 __sip_ack: Stopping retransmission on '5d28f46f54fd450475a9f02419f99674@24.123.23.170' of Request 102: Match Found
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:9642 handle_response_invite: SIP response 200 to standard invite

[Mar 10 12:02:05] Found RTP audio format 0

[Mar 10 12:02:05] Found RTP audio format 101

[Mar 10 12:02:05] Peer audio RTP is at port 192.168.1.85:30012
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3631 process_sdp: Peer audio RTP is at port 192.168.1.85:30012

[Mar 10 12:02:05] Peer video RTP is at port 192.168.1.85:65535
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3653 process_sdp: Peer video RTP is at port 192.168.1.85:65535

[Mar 10 12:02:05] Found description format PCMU

[Mar 10 12:02:05] Found description format telephone-event

[Mar 10 12:02:05] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)

[Mar 10 12:02:05] Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:6194 build_route: build_route: Contact hop: <sip:523@192.168.1.85:5060>

[Mar 10 12:02:05] list_route: hop: <sip:523@192.168.1.85:5060>

[Mar 10 12:02:05] set_destination: Parsing <sip:523@192.168.1.85:5060> for address/port to send to

[Mar 10 12:02:05] set_destination: set destination to 192.168.1.85, port 5060

[Mar 10 12:02:05] Transmitting (no NAT) to 192.168.1.85:5060:
ACK sip:523@192.168.1.85:5060 SIP/2.0
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK58559bfb
From: "593 593" <sip:593@24.123.23.170>;tag=as0da2dd59
To: <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f
Contact: <sip:593@24.123.23.170>
Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---

[Mar 10 12:02:05]     -- SIP/523-0850b038 answered SIP/593-084f44b8
[Mar 10 12:02:05] DEBUG[15195]: chan_sip.c:2557 sip_answer: sip_answer(SIP/593-084f44b8)

[Mar 10 12:02:05] We're at 24.123.23.170 port 19424

[Mar 10 12:02:05] Video is at 24.123.23.170 port 13400

[Mar 10 12:02:05] Adding codec 0x4 (ulaw) to SDP

[Mar 10 12:02:05] Adding codec 0x8 (alaw) to SDP

[Mar 10 12:02:05] Adding non-codec 0x1 (telephone-event) to SDP

[Mar 10 12:02:05] Reliably Transmitting (no NAT) to 192.168.1.83:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.83:5060;branch=z9hG4bKe316288f6;received=192.168.1.83
From: 593 <sip:593@24.123.23.170:5060>;tag=b3eeb4c9225baf5
To: 523 <sip:523@24.123.23.170:5060>;tag=as38239579
Call-ID: 20a99672af308312d36264481f6b7f5a@192.168.1.83
CSeq: 75546811 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:523@24.123.23.170>
Content-Type: application/sdp
Content-Length: 242

v=0
o=root 15131 15131 IN IP4 24.123.23.170
s=session
c=IN IP4 24.123.23.170
t=0 0
m=audio 19424 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
[Mar 10 12:02:05] DEBUG[15195]: chan_sip.c:1304 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id  #109

[Mar 10 12:02:05]     -- Attempting native bridge of SIP/593-084f44b8 and SIP/523-0850b038
[Mar 10 12:02:05] DEBUG[15195]: chan_sip.c:13034 sip_set_rtp_peer: Deferring reinvite on SIP '20a99672af308312d36264481f6b7f5a@192.168.1.83' - It's audio will be redirected to IP 192.168.1.85
[Mar 10 12:02:05] DEBUG[15195]: chan_sip.c:13028 sip_set_rtp_peer: Sending reinvite on SIP '5d28f46f54fd450475a9f02419f99674@24.123.23.170' - It's audio soon redirected to IP 192.168.1.83

[Mar 10 12:02:05] set_destination: Parsing <sip:523@192.168.1.85:5060> for address/port to send to

[Mar 10 12:02:05] set_destination: set destination to 192.168.1.85, port 5060

[Mar 10 12:02:05] We're at 24.123.23.170 port 17986

[Mar 10 12:02:05] Video is at 24.123.23.170 port 12600

[Mar 10 12:02:05] Adding codec 0x4 (ulaw) to SDP

[Mar 10 12:02:05] Adding codec 0x8 (alaw) to SDP

[Mar 10 12:02:05] Adding codec 0x100 (g729) to SDP

[Mar 10 12:02:05] Adding non-codec 0x1 (telephone-event) to SDP
[Mar 10 12:02:05] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 0: INVITE sip:523@192.168.1.85:5060 SIP/2.0 (40)
[Mar 10 12:02:05] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5e7522ff (58)
[Mar 10 12:02:05] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 2: From: "593 593" <sip:593@24.123.23.170>;tag=as0da2dd59 (54)
[Mar 10 12:02:05] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 3: To: <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f (68)
[Mar 10 12:02:05] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 4: Contact: <sip:593@24.123.23.170> (32)
[Mar 10 12:02:05] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 5: Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170 (55)
[Mar 10 12:02:05] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 6: CSeq: 103 INVITE (16)
[Mar 10 12:02:05] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 7: User-Agent: Asterisk PBX (24)
[Mar 10 12:02:05] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 8: Max-Forwards: 70 (16)
[Mar 10 12:02:05] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66)
[Mar 10 12:02:05] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 10: X-asterisk-info: SIP re-invite (RTP bridge) (43)
[Mar 10 12:02:05] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 11: Content-Type: application/sdp (29)
[Mar 10 12:02:05] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 12: Content-Length: 287 (19)
[Mar 10 12:02:05] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 13:  (0)
[Mar 10 12:02:05] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: v=0 (3)
[Mar 10 12:02:05] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: o=root 15131 15132 IN IP4 192.168.1.83 (38)
[Mar 10 12:02:05] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: s=session (9)
[Mar 10 12:02:05] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: c=IN IP4 192.168.1.83 (21)
[Mar 10 12:02:05] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: t=0 0 (5)
[Mar 10 12:02:05] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: m=audio 51000 RTP/AVP 0 8 18 101 (32)
[Mar 10 12:02:05] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: a=rtpmap:0 PCMU/8000 (20)
[Mar 10 12:02:05] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: a=rtpmap:8 PCMA/8000 (20)
[Mar 10 12:02:05] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: a=rtpmap:18 G729/8000 (21)
[Mar 10 12:02:05] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: a=fmtp:18 annexb=no (19)
[Mar 10 12:02:05] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33)
[Mar 10 12:02:05] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: a=fmtp:101 0-16 (15)
[Mar 10 12:02:05] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: a=silenceSupp:off - - - - (25)

[Mar 10 12:02:05] 13 headers, 13 lines

[Mar 10 12:02:05] Reliably Transmitting (no NAT) to 192.168.1.85:5060:
INVITE sip:523@192.168.1.85:5060 SIP/2.0
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5e7522ff
From: "593 593" <sip:593@24.123.23.170>;tag=as0da2dd59
To: <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f
Contact: <sip:593@24.123.23.170>
Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 15131 15132 IN IP4 192.168.1.83
s=session
c=IN IP4 192.168.1.83
t=0 0
m=audio 51000 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
[Mar 10 12:02:05] DEBUG[15195]: chan_sip.c:1304 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id  #110
[Mar 10 12:02:05] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 523
[Mar 10 12:02:05] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/523 - state 2 (In use)
[Mar 10 12:02:05] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 593
[Mar 10 12:02:05] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/593 - state 2 (In use)
[Mar 10 12:02:05] DEBUG[15199]: app_queue.c:500 changethread: Device 'SIP/523' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[Mar 10 12:02:05] DEBUG[15200]: app_queue.c:500 changethread: Device 'SIP/593' changed to state '2' (In use) but we don't care because they're not a member of any queue.

[Mar 10 12:02:05] 
<-- SIP read from 192.168.1.83:5060: 
ACK sip:523@24.123.23.170 SIP/2.0
Max-Forwards: 70
Content-Length: 0
Via: SIP/2.0/UDP 192.168.1.83:5060;branch=z9hG4bKe8f8f0d2c
Call-ID: 20a99672af308312d36264481f6b7f5a@192.168.1.83
From: 593 <sip:593@24.123.23.170:5060>;tag=b3eeb4c9225baf5
To: 523 <sip:523@24.123.23.170:5060>;tag=as38239579
CSeq: 75546811 ACK
Proxy-Authorization:Digest response="f2ad6f8c0955a18b6c8fd9a6bf574bdf",username="593",realm="asterisk",nonce="4de95353",algorithm=MD5,uri="sip:523@24.123.23.170:5060"
User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4


[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: ACK sip:523@24.123.23.170 SIP/2.0 (33)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Max-Forwards: 70 (16)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: Content-Length: 0 (17)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: Via: SIP/2.0/UDP 192.168.1.83:5060;branch=z9hG4bKe8f8f0d2c (58)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Call-ID: 20a99672af308312d36264481f6b7f5a@192.168.1.83 (54)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: From: 593 <sip:593@24.123.23.170:5060>;tag=b3eeb4c9225baf5 (58)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: To: 523 <sip:523@24.123.23.170:5060>;tag=as38239579 (51)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: CSeq: 75546811 ACK (18)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Proxy-Authorization:Digest response="f2ad6f8c0955a18b6c8fd9a6bf574bdf",username="593",realm="asterisk",nonce="4de95353",algorithm=MD5,uri="sip:523@24.123.23.170:5060" (166)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 (42)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10:  (0)

[Mar 10 12:02:05] --- (10 headers 0 lines) ---
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received ACK (6) - Command in SIP ACK
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:1401 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #109
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:1412 __sip_ack: Stopping retransmission on '20a99672af308312d36264481f6b7f5a@192.168.1.83' of Response 75546811: Match Found
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:9625 check_pendings: Sending pending reinvite on '20a99672af308312d36264481f6b7f5a@192.168.1.83'

[Mar 10 12:02:05] set_destination: Parsing <sip:593@192.168.1.83:5060> for address/port to send to

[Mar 10 12:02:05] set_destination: set destination to 192.168.1.83, port 5060

[Mar 10 12:02:05] We're at 24.123.23.170 port 19424

[Mar 10 12:02:05] Video is at 24.123.23.170 port 13400

[Mar 10 12:02:05] Adding codec 0x4 (ulaw) to SDP

[Mar 10 12:02:05] Adding non-codec 0x1 (telephone-event) to SDP
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: INVITE sip:593@192.168.1.83:5060 SIP/2.0 (40)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK4a8d437d (58)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: From: 523 <sip:523@24.123.23.170:5060>;tag=as38239579 (53)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: To: 593 <sip:593@24.123.23.170:5060>;tag=b3eeb4c9225baf5 (56)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Contact: <sip:523@24.123.23.170> (32)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: Call-ID: 20a99672af308312d36264481f6b7f5a@192.168.1.83 (54)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: CSeq: 102 INVITE (16)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: User-Agent: Asterisk PBX (24)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Max-Forwards: 70 (16)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: X-asterisk-info: SIP re-invite (RTP bridge) (43)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11: Content-Type: application/sdp (29)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 12: Content-Length: 216 (19)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 13:  (0)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: v=0 (3)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: o=root 15131 15132 IN IP4 192.168.1.85 (38)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: s=session (9)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: c=IN IP4 192.168.1.85 (21)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: t=0 0 (5)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: m=audio 30012 RTP/AVP 0 101 (27)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:0 PCMU/8000 (20)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=fmtp:101 0-16 (15)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=silenceSupp:off - - - - (25)

[Mar 10 12:02:05] 13 headers, 10 lines

[Mar 10 12:02:05] Reliably Transmitting (no NAT) to 192.168.1.83:5060:
INVITE sip:593@192.168.1.83:5060 SIP/2.0
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK4a8d437d
From: 523 <sip:523@24.123.23.170:5060>;tag=as38239579
To: 593 <sip:593@24.123.23.170:5060>;tag=b3eeb4c9225baf5
Contact: <sip:523@24.123.23.170>
Call-ID: 20a99672af308312d36264481f6b7f5a@192.168.1.83
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 216

v=0
o=root 15131 15132 IN IP4 192.168.1.85
s=session
c=IN IP4 192.168.1.85
t=0 0
m=audio 30012 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:1304 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id  #111

[Mar 10 12:02:05] 
<-- SIP read from 192.168.1.97:5060: 


[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0:  (0)

[Mar 10 12:02:05] --- (0 headers 0 lines) Nat keepalive ---

[Mar 10 12:02:05] 
<-- SIP read from 192.168.1.85:5060: 
SIP/2.0 200 OK
To: <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f
Contact: <sip:523@192.168.1.85:5060>
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5e7522ff
From: "593 593" <sip:593@24.123.23.170>;tag=as0da2dd59
Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170
CSeq: 103 INVITE
Content-Type: application/sdp
Content-Length: 197
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK

v=0
o=- 15131 248056 IN IP4 192.168.1.85
s=session
c=IN IP4 192.168.1.85
t=0 0
m=audio 30012 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -

[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: SIP/2.0 200 OK (14)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: To: <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f (68)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: Contact: <sip:523@192.168.1.85:5060> (36)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5e7522ff (58)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: From: "593 593" <sip:593@24.123.23.170>;tag=as0da2dd59 (54)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170 (55)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: CSeq: 103 INVITE (16)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Content-Type: application/sdp (29)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Content-Length: 197 (19)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Supported: sip-cc, sip-cc-01, replaces, timer (45)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11:  (0)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: v=0 (3)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: o=- 15131 248056 IN IP4 192.168.1.85 (36)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: s=session (9)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: c=IN IP4 192.168.1.85 (21)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: t=0 0 (5)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: m=audio 30012 RTP/AVP 0 101 (27)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:0 PCMU/8000 (20)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=silenceSupp:off - - - - (25)

[Mar 10 12:02:05] --- (11 headers 9 lines) ---
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:1390 __sip_ack: Acked pending invite 103
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:1401 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #110
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:1412 __sip_ack: Stopping retransmission on '5d28f46f54fd450475a9f02419f99674@24.123.23.170' of Request 103: Match Found
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:9640 handle_response_invite: SIP response 200 to RE-invite on outgoing call 5d28f46f54fd450475a9f02419f99674@24.123.23.170

[Mar 10 12:02:05] Found RTP audio format 0

[Mar 10 12:02:05] Found RTP audio format 101

[Mar 10 12:02:05] Peer audio RTP is at port 192.168.1.85:30012
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3631 process_sdp: Peer audio RTP is at port 192.168.1.85:30012

[Mar 10 12:02:05] Peer video RTP is at port 192.168.1.85:65535
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3653 process_sdp: Peer video RTP is at port 192.168.1.85:65535

[Mar 10 12:02:05] Found description format PCMU

[Mar 10 12:02:05] Found description format telephone-event

[Mar 10 12:02:05] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)

[Mar 10 12:02:05] Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:6137 build_route: build_route: Retaining previous route: <sip:523@192.168.1.85:5060>

[Mar 10 12:02:05] set_destination: Parsing <sip:523@192.168.1.85:5060> for address/port to send to

[Mar 10 12:02:05] set_destination: set destination to 192.168.1.85, port 5060

[Mar 10 12:02:05] Transmitting (no NAT) to 192.168.1.85:5060:
ACK sip:523@192.168.1.85:5060 SIP/2.0
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK204db7bd
From: "593 593" <sip:593@24.123.23.170>;tag=as0da2dd59
To: <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f
Contact: <sip:593@24.123.23.170>
Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---

[Mar 10 12:02:05] 
<-- SIP read from 192.168.1.83:5060: 
SIP/2.0 200 OK
Call-ID: 20a99672af308312d36264481f6b7f5a@192.168.1.83
CSeq: 102 INVITE
From: 523 <sip:523@24.123.23.170:5060>;tag=as38239579
To: 593 <sip:593@24.123.23.170:5060>;tag=b3eeb4c9225baf5
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK4a8d437d
Content-Length: 230
Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE
Content-Type: application/sdp
Supported: replaces
Contact: 593 <sip:593@192.168.1.83:5060>
User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4

v=0
o=MxSIP 0 1158539950 IN IP4 192.168.1.83
s=SIP Call
c=IN IP4 192.168.1.83
t=0 0
m=audio 51000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:on - - - -

[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: SIP/2.0 200 OK (14)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Call-ID: 20a99672af308312d36264481f6b7f5a@192.168.1.83 (54)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: CSeq: 102 INVITE (16)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: From: 523 <sip:523@24.123.23.170:5060>;tag=as38239579 (53)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: To: 593 <sip:593@24.123.23.170:5060>;tag=b3eeb4c9225baf5 (56)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK4a8d437d (58)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: Content-Length: 230 (19)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE (55)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Content-Type: application/sdp (29)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Supported: replaces (19)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Contact: 593 <sip:593@192.168.1.83:5060> (40)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11: User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 (42)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 12:  (0)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: v=0 (3)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: o=MxSIP 0 1158539950 IN IP4 192.168.1.83 (40)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: s=SIP Call (10)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: c=IN IP4 192.168.1.83 (21)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: t=0 0 (5)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: m=audio 51000 RTP/AVP 0 101 (27)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:0 PCMU/8000 (20)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=fmtp:101 0-15 (15)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=ptime:20 (10)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=silenceSupp:on - - - - (24)

[Mar 10 12:02:05] --- (12 headers 11 lines) ---
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:1390 __sip_ack: Acked pending invite 102
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:1401 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #111
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:1412 __sip_ack: Stopping retransmission on '20a99672af308312d36264481f6b7f5a@192.168.1.83' of Request 102: Match Found
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:9640 handle_response_invite: SIP response 200 to RE-invite on outgoing call 20a99672af308312d36264481f6b7f5a@192.168.1.83

[Mar 10 12:02:05] Found RTP audio format 0

[Mar 10 12:02:05] Found RTP audio format 101

[Mar 10 12:02:05] Peer audio RTP is at port 192.168.1.83:51000
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3631 process_sdp: Peer audio RTP is at port 192.168.1.83:51000

[Mar 10 12:02:05] Peer video RTP is at port 192.168.1.83:65535
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:3653 process_sdp: Peer video RTP is at port 192.168.1.83:65535

[Mar 10 12:02:05] Found description format PCMU

[Mar 10 12:02:05] Found description format telephone-event

[Mar 10 12:02:05] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)

[Mar 10 12:02:05] Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Mar 10 12:02:05] DEBUG[15149]: chan_sip.c:6194 build_route: build_route: Contact hop: 593 <sip:593@192.168.1.83:5060>

[Mar 10 12:02:05] list_route: hop: <sip:593@192.168.1.83:5060>

[Mar 10 12:02:05] set_destination: Parsing <sip:593@192.168.1.83:5060> for address/port to send to

[Mar 10 12:02:05] set_destination: set destination to 192.168.1.83, port 5060

[Mar 10 12:02:05] Transmitting (no NAT) to 192.168.1.83:5060:
ACK sip:593@192.168.1.83:5060 SIP/2.0
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5ab05639
From: 523 <sip:523@24.123.23.170:5060>;tag=as38239579
To: 593 <sip:593@24.123.23.170:5060>;tag=b3eeb4c9225baf5
Contact: <sip:523@24.123.23.170>
Call-ID: 20a99672af308312d36264481f6b7f5a@192.168.1.83
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
[Mar 10 12:02:07] NOTICE[15149]: chan_sip.c:5399 sip_reregister:    -- Re-registration for  3173241052@sip.broadvoice.com
[Mar 10 12:02:07] DEBUG[15149]: chan_sip.c:3168 sip_alloc: Allocating new SIP dialog for 71f439ca643c96aa63da468e6d077ab4@24.123.23.170 - REGISTER (No RTP)
[Mar 10 12:02:07] DEBUG[15149]: chan_sip.c:5564 transmit_register: Scheduled a registration timeout for sip.broadvoice.com id  #112 
[Mar 10 12:02:07] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: REGISTER sip:sip.broadvoice.com SIP/2.0 (39)
[Mar 10 12:02:07] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK215c0c90;rport (64)
[Mar 10 12:02:07] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: From: <sip:3173241052@sip.broadvoice.com>;tag=as11f5452a (56)
[Mar 10 12:02:07] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: To: <sip:3173241052@sip.broadvoice.com> (39)
[Mar 10 12:02:07] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Call-ID: 71f439ca643c96aa63da468e6d077ab4@24.123.23.170 (55)
[Mar 10 12:02:07] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: CSeq: 104 REGISTER (18)
[Mar 10 12:02:07] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: User-Agent: Asterisk PBX (24)
[Mar 10 12:02:07] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Max-Forwards: 70 (16)
[Mar 10 12:02:07] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Expires: 120 (12)
[Mar 10 12:02:07] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Contact: <sip:3173241052@24.123.23.170> (39)
[Mar 10 12:02:07] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Event: registration (19)
[Mar 10 12:02:07] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11: Content-Length: 0 (17)
[Mar 10 12:02:07] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 12:  (0)

[Mar 10 12:02:07] REGISTER 12 headers, 0 lines

[Mar 10 12:02:07] REGISTER attempt 1 to 3173241052@sip.broadvoice.com

[Mar 10 12:02:07] Reliably Transmitting (no NAT) to 147.135.12.128:5060:
REGISTER sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK215c0c90;rport
From: <sip:3173241052@sip.broadvoice.com>;tag=as11f5452a
To: <sip:3173241052@sip.broadvoice.com>
Call-ID: 71f439ca643c96aa63da468e6d077ab4@24.123.23.170
CSeq: 104 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: <sip:3173241052@24.123.23.170>
Event: registration
Content-Length: 0


---
[Mar 10 12:02:07] DEBUG[15149]: chan_sip.c:1304 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id  #113

[Mar 10 12:02:07] 
<-- SIP read from 147.135.12.128:5060: 
SIP/2.0 200 OK
Call-ID: 71f439ca643c96aa63da468e6d077ab4@24.123.23.170
CSeq: 104 REGISTER
From: <sip:3173241052@sip.broadvoice.com>;tag=as11f5452a
To: <sip:3173241052@sip.broadvoice.com>
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK215c0c90
Contact: <sip:3173241052@24.123.23.170>
Expires: 30
Event: registration
Content-Length:    0


[Mar 10 12:02:07] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: SIP/2.0 200 OK (14)
[Mar 10 12:02:07] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Call-ID: 71f439ca643c96aa63da468e6d077ab4@24.123.23.170 (55)
[Mar 10 12:02:07] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: CSeq: 104 REGISTER (18)
[Mar 10 12:02:07] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: From: <sip:3173241052@sip.broadvoice.com>;tag=as11f5452a (56)
[Mar 10 12:02:07] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: To: <sip:3173241052@sip.broadvoice.com> (39)
[Mar 10 12:02:07] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK215c0c90 (58)
[Mar 10 12:02:07] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: Contact: <sip:3173241052@24.123.23.170> (39)
[Mar 10 12:02:07] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Expires: 30 (11)
[Mar 10 12:02:07] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Event: registration (19)
[Mar 10 12:02:07] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Content-Length:    0 (20)
[Mar 10 12:02:07] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10:  (0)

[Mar 10 12:02:07] --- (10 headers 0 lines) ---
[Mar 10 12:02:07] DEBUG[15149]: chan_sip.c:1401 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #113
[Mar 10 12:02:07] DEBUG[15149]: chan_sip.c:1412 __sip_ack: Stopping retransmission on '71f439ca643c96aa63da468e6d077ab4@24.123.23.170' of Request 104: Match Found
[Mar 10 12:02:07] DEBUG[15149]: chan_sip.c:9843 handle_response_register: Registration successful
[Mar 10 12:02:07] DEBUG[15149]: chan_sip.c:9845 handle_response_register: Cancelling timeout 112

[Mar 10 12:02:07] Scheduling destruction of call '71f439ca643c96aa63da468e6d077ab4@24.123.23.170' in 32000 ms
[Mar 10 12:02:07] NOTICE[15149]: chan_sip.c:9895 handle_response_register: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 23 s)

[Mar 10 12:02:11] 
<-- SIP read from 192.168.1.97:5060: 
INVITE sip:523@24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.97:5060;rport;branch=z9hG4bK412717690
From: "Display Name" <sip:551@24.123.23.170>;tag=1341134544
To: <sip:523@24.123.23.170>
Call-ID: 3175191617@192.168.1.97
CSeq: 1000 INVITE
Contact: <sip:551@192.168.1.97:5060>
max-forwards: 70
user-agent: UTSTARCOM F1000/Device ID-0007ba26174b
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, MESSAGE, REGISTER, REFER, PING, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 298  

v=0
o=551 123456 654329 IN IP4 192.168.1.97
s=none
c=IN IP4 192.168.1.97
t=0 0
m=audio 10010 RTP/AVP 0 8 18 2 101
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729A/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: INVITE sip:523@24.123.23.170 SIP/2.0 (36)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.97:5060;rport;branch=z9hG4bK412717690 (64)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: From: "Display Name" <sip:551@24.123.23.170>;tag=1341134544 (59)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: To: <sip:523@24.123.23.170> (27)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Call-ID: 3175191617@192.168.1.97 (32)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: CSeq: 1000 INVITE (17)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: Contact: <sip:551@192.168.1.97:5060> (36)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: max-forwards: 70 (16)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: user-agent: UTSTARCOM F1000/Device ID-0007ba26174b (50)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, MESSAGE, REGISTER, REFER, PING, NOTIFY, INFO (86)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Content-Type: application/sdp (29)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11: Content-Length: 298   (21)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 12:  (0)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: v=0 (3)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: o=551 123456 654329 IN IP4 192.168.1.97 (39)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: s=none (6)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: c=IN IP4 192.168.1.97 (21)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: t=0 0 (5)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: m=audio 10010 RTP/AVP 0 8 18 2 101 (34)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=ptime:20 (10)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:0 PCMU/8000 (20)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:8 PCMA/8000 (20)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:18 G729A/8000 (22)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=fmtp:18 annexb=no (19)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:2 G726-32/8000 (23)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=fmtp:101 0-15 (15)

[Mar 10 12:02:11] --- (12 headers 14 lines) ---
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3168 sip_alloc: Allocating new SIP dialog for 3175191617@192.168.1.97 - INVITE (With RTP)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received INVITE (5) - Command in SIP INVITE

[Mar 10 12:02:11] Using INVITE request as basis request - 3175191617@192.168.1.97

[Mar 10 12:02:11] Sending to 192.168.1.97 : 5060 (NAT)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:7215 check_user_full: Setting NAT on RTP to 524288
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:7219 check_user_full: Setting NAT on VRTP to 524288

[Mar 10 12:02:11] Reliably Transmitting (NAT) to 192.168.1.97:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.97:5060;branch=z9hG4bK412717690;received=192.168.1.97;rport=5060
From: "Display Name" <sip:551@24.123.23.170>;tag=1341134544
To: <sip:523@24.123.23.170>;tag=as535de7d7
Call-ID: 3175191617@192.168.1.97
CSeq: 1000 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="38387c0e"
Content-Length: 0


---
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:1304 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id  #116

[Mar 10 12:02:11] Scheduling destruction of call '3175191617@192.168.1.97' in 15000 ms

[Mar 10 12:02:11] Found user '551'

[Mar 10 12:02:11] 
<-- SIP read from 192.168.1.97:5060: 
ACK sip:523@24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.97:5060;rport;branch=z9hG4bK412717690
From: "Display Name" <sip:551@24.123.23.170>;tag=1341134544
To: <sip:523@24.123.23.170>;tag=as535de7d7
Call-ID: 3175191617@192.168.1.97
CSeq: 1000 ACK
Content-Length: 0


[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: ACK sip:523@24.123.23.170 SIP/2.0 (33)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.97:5060;rport;branch=z9hG4bK412717690 (64)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: From: "Display Name" <sip:551@24.123.23.170>;tag=1341134544 (59)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: To: <sip:523@24.123.23.170>;tag=as535de7d7 (42)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Call-ID: 3175191617@192.168.1.97 (32)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: CSeq: 1000 ACK (14)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: Content-Length: 0 (17)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7:  (0)

[Mar 10 12:02:11] --- (7 headers 0 lines) ---
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received ACK (6) - Command in SIP ACK
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:1401 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #116
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:1412 __sip_ack: Stopping retransmission on '3175191617@192.168.1.97' of Response 1000: Match Found

[Mar 10 12:02:11] 
<-- SIP read from 192.168.1.97:5060: 
INVITE sip:523@24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.97:5060;rport;branch=z9hG4bK3702870459
From: "Display Name" <sip:551@24.123.23.170>;tag=1341134544
To: <sip:523@24.123.23.170>
Call-ID: 3175191617@192.168.1.97
CSeq: 1001 INVITE
Contact: <sip:551@192.168.1.97:5060>
Proxy-Authorization: Digest username="551", realm="asterisk", nonce="38387c0e", uri="sip:523@24.123.23.170", response="7a4570e5a58b5afad9ad20bcdb87bb4a", algorithm=MD5
max-forwards: 70
user-agent: UTSTARCOM F1000/Device ID-0007ba26174b
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, MESSAGE, REGISTER, REFER, PING, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 298  

v=0
o=551 123456 654329 IN IP4 192.168.1.97
s=none
c=IN IP4 192.168.1.97
t=0 0
m=audio 10010 RTP/AVP 0 8 18 2 101
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729A/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: INVITE sip:523@24.123.23.170 SIP/2.0 (36)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.97:5060;rport;branch=z9hG4bK3702870459 (65)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: From: "Display Name" <sip:551@24.123.23.170>;tag=1341134544 (59)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: To: <sip:523@24.123.23.170> (27)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Call-ID: 3175191617@192.168.1.97 (32)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: CSeq: 1001 INVITE (17)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: Contact: <sip:551@192.168.1.97:5060> (36)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Proxy-Authorization: Digest username="551", realm="asterisk", nonce="38387c0e", uri="sip:523@24.123.23.170", response="7a4570e5a58b5afad9ad20bcdb87bb4a", algorithm=MD5 (167)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: max-forwards: 70 (16)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: user-agent: UTSTARCOM F1000/Device ID-0007ba26174b (50)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, MESSAGE, REGISTER, REFER, PING, NOTIFY, INFO (86)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11: Content-Type: application/sdp (29)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 12: Content-Length: 298   (21)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 13:  (0)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: v=0 (3)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: o=551 123456 654329 IN IP4 192.168.1.97 (39)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: s=none (6)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: c=IN IP4 192.168.1.97 (21)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: t=0 0 (5)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: m=audio 10010 RTP/AVP 0 8 18 2 101 (34)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=ptime:20 (10)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:0 PCMU/8000 (20)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:8 PCMA/8000 (20)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:18 G729A/8000 (22)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=fmtp:18 annexb=no (19)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:2 G726-32/8000 (23)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=fmtp:101 0-15 (15)

[Mar 10 12:02:11] --- (13 headers 14 lines) ---
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received INVITE (5) - Command in SIP INVITE

[Mar 10 12:02:11] Using INVITE request as basis request - 3175191617@192.168.1.97

[Mar 10 12:02:11] Sending to 192.168.1.97 : 5060 (NAT)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:7215 check_user_full: Setting NAT on RTP to 524288
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:7219 check_user_full: Setting NAT on VRTP to 524288

[Mar 10 12:02:11] Found user '551'

[Mar 10 12:02:11] Found RTP audio format 0

[Mar 10 12:02:11] Found RTP audio format 8

[Mar 10 12:02:11] Found RTP audio format 18

[Mar 10 12:02:11] Found RTP audio format 2

[Mar 10 12:02:11] Found RTP audio format 101

[Mar 10 12:02:11] Peer audio RTP is at port 192.168.1.97:10010
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3631 process_sdp: Peer audio RTP is at port 192.168.1.97:10010

[Mar 10 12:02:11] Peer video RTP is at port 192.168.1.97:65535
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3653 process_sdp: Peer video RTP is at port 192.168.1.97:65535

[Mar 10 12:02:11] Found description format PCMU

[Mar 10 12:02:11] Found description format PCMA

[Mar 10 12:02:11] Found description format G729A

[Mar 10 12:02:11] Found description format G726-32

[Mar 10 12:02:11] Found description format telephone-event

[Mar 10 12:02:11] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x11c (ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)

[Mar 10 12:02:11] Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:10571 handle_request_invite: Checking SIP call limits for device 551
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:2217 update_call_counter: Updating call counter for incoming call

[Mar 10 12:02:11] Looking for 523 in smvoice-sip (domain 24.123.23.170)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:6194 build_route: build_route: Contact hop: <sip:551@192.168.1.97:5060>

[Mar 10 12:02:11] list_route: hop: <sip:551@192.168.1.97:5060>

[Mar 10 12:02:11] Transmitting (NAT) to 192.168.1.97:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.97:5060;branch=z9hG4bK3702870459;received=192.168.1.97;rport=5060
From: "Display Name" <sip:551@24.123.23.170>;tag=1341134544
To: <sip:523@24.123.23.170>
Call-ID: 3175191617@192.168.1.97
CSeq: 1001 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:523@24.123.23.170>
Content-Length: 0


---
[Mar 10 12:02:11] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 551
[Mar 10 12:02:11] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/551 - state 2 (In use)
[Mar 10 12:02:11] DEBUG[15201]: pbx.c:1677 pbx_extension_helper: Launching 'NoOp'

[Mar 10 12:02:11]     -- Executing NoOp("SIP/551-08518cd8", "5xx") in new stack
[Mar 10 12:02:11] DEBUG[15201]: pbx.c:1677 pbx_extension_helper: Launching 'Set'

[Mar 10 12:02:11]     -- Executing Set("SIP/551-08518cd8", "SMVOICE_CONTEXT_EXTEN=523") in new stack
[Mar 10 12:02:11] DEBUG[15201]: pbx.c:1677 pbx_extension_helper: Launching 'AGI'

[Mar 10 12:02:11]     -- Executing AGI("SIP/551-08518cd8", "smvoice|-company|demonstration|-digium_asterisk|-asterisk_callat_forwarding|523") in new stack

[Mar 10 12:02:11]     -- Launched AGI Script /var/lib/asterisk/agi-bin/smvoice
[Mar 10 12:02:11] DEBUG[15202]: app_queue.c:500 changethread: Device 'SIP/551' changed to state '2' (In use) but we don't care because they're not a member of any queue.

[Mar 10 12:02:11]     -- AGI Script smvoice completed, returning 0
[Mar 10 12:02:11] DEBUG[15201]: pbx.c:1589 pbx_substitute_variables_helper_full: Expression result is '0'
[Mar 10 12:02:11] DEBUG[15201]: pbx.c:1677 pbx_extension_helper: Launching 'GotoIf'

[Mar 10 12:02:11]     -- Executing GotoIf("SIP/551-08518cd8", "0?INVALID|1") in new stack
[Mar 10 12:02:11] DEBUG[15201]: pbx.c:6177 pbx_builtin_gotoif: Not taking any branch
[Mar 10 12:02:11] DEBUG[15201]: pbx.c:1589 pbx_substitute_variables_helper_full: Expression result is '0'
[Mar 10 12:02:11] DEBUG[15201]: pbx.c:1677 pbx_extension_helper: Launching 'GotoIf'

[Mar 10 12:02:11]     -- Executing GotoIf("SIP/551-08518cd8", "0?_5XX-NOANSWER|1") in new stack
[Mar 10 12:02:11] DEBUG[15201]: pbx.c:6177 pbx_builtin_gotoif: Not taking any branch
[Mar 10 12:02:11] DEBUG[15201]: pbx.c:1677 pbx_extension_helper: Launching 'Dial'

[Mar 10 12:02:11]     -- Executing Dial("SIP/551-08518cd8", "SIP/523|20|") in new stack
[Mar 10 12:02:11] DEBUG[15201]: chan_sip.c:3168 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP)
[Mar 10 12:02:11] DEBUG[15201]: chan_sip.c:1885 create_addr_from_peer: Setting NAT on RTP to 0
[Mar 10 12:02:11] DEBUG[15201]: chan_sip.c:1889 create_addr_from_peer: Setting NAT on VRTP to 0
[Mar 10 12:02:11] DEBUG[15201]: channel.c:2902 ast_channel_inherit_variables: Not copying variable STACK-smvoice-sip-523-6.
[Mar 10 12:02:11] DEBUG[15201]: channel.c:2902 ast_channel_inherit_variables: Not copying variable STACK-smvoice-sip-523-5.
[Mar 10 12:02:11] DEBUG[15201]: channel.c:2902 ast_channel_inherit_variables: Not copying variable STACK-smvoice-sip-523-4.
[Mar 10 12:02:11] DEBUG[15201]: channel.c:2902 ast_channel_inherit_variables: Not copying variable SMVOICE_WORK_EXTENSION.
[Mar 10 12:02:11] DEBUG[15201]: channel.c:2902 ast_channel_inherit_variables: Not copying variable SMVOICE_EXTENSION.
[Mar 10 12:02:11] DEBUG[15201]: channel.c:2902 ast_channel_inherit_variables: Not copying variable SMVOICE_PIN.
[Mar 10 12:02:11] DEBUG[15201]: channel.c:2902 ast_channel_inherit_variables: Not copying variable SMVOICE_CALLAT.
[Mar 10 12:02:11] DEBUG[15201]: channel.c:2902 ast_channel_inherit_variables: Not copying variable STACK-smvoice-sip-523-3.
[Mar 10 12:02:11] DEBUG[15201]: channel.c:2902 ast_channel_inherit_variables: Not copying variable SMVOICE_CONTEXT_EXTEN.
[Mar 10 12:02:11] DEBUG[15201]: channel.c:2902 ast_channel_inherit_variables: Not copying variable STACK-smvoice-sip-523-2.
[Mar 10 12:02:11] DEBUG[15201]: channel.c:2902 ast_channel_inherit_variables: Not copying variable STACK-smvoice-sip-523-1.
[Mar 10 12:02:11] DEBUG[15201]: channel.c:2902 ast_channel_inherit_variables: Not copying variable SIPCALLID.
[Mar 10 12:02:11] DEBUG[15201]: channel.c:2902 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT.
[Mar 10 12:02:11] DEBUG[15201]: channel.c:2902 ast_channel_inherit_variables: Not copying variable SIPDOMAIN.
[Mar 10 12:02:11] DEBUG[15201]: channel.c:2902 ast_channel_inherit_variables: Not copying variable SIPURI.
[Mar 10 12:02:11] DEBUG[15201]: chan_sip.c:2079 sip_call: Outgoing Call for 523
[Mar 10 12:02:11] DEBUG[15201]: chan_sip.c:2217 update_call_counter: Updating call counter for outgoing call

[Mar 10 12:02:11] We're at 24.123.23.170 port 14572

[Mar 10 12:02:11] Video is at 24.123.23.170 port 16640

[Mar 10 12:02:11] Adding codec 0x4 (ulaw) to SDP

[Mar 10 12:02:11] Adding codec 0x8 (alaw) to SDP

[Mar 10 12:02:11] Adding codec 0x2 (gsm) to SDP

[Mar 10 12:02:11] Adding non-codec 0x1 (telephone-event) to SDP
[Mar 10 12:02:11] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 0: INVITE sip:523@192.168.1.85:5060 SIP/2.0 (40)
[Mar 10 12:02:11] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK202ed764 (58)
[Mar 10 12:02:11] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 2: From: "551 551" <sip:551@24.123.23.170>;tag=as6fedd0f9 (54)
[Mar 10 12:02:11] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 3: To: <sip:523@192.168.1.85:5060> (31)
[Mar 10 12:02:11] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 4: Contact: <sip:551@24.123.23.170> (32)
[Mar 10 12:02:11] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 5: Call-ID: 3890f17d116db7343bb188d42c093fd2@24.123.23.170 (55)
[Mar 10 12:02:11] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 6: CSeq: 102 INVITE (16)
[Mar 10 12:02:11] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 7: User-Agent: Asterisk PBX (24)
[Mar 10 12:02:11] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 8: Max-Forwards: 70 (16)
[Mar 10 12:02:11] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 9: Date: Sat, 10 Mar 2007 17:02:11 GMT (35)
[Mar 10 12:02:11] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66)
[Mar 10 12:02:11] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 11: Content-Type: application/sdp (29)
[Mar 10 12:02:11] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 12: Content-Length: 265 (19)
[Mar 10 12:02:11] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 13:  (0)
[Mar 10 12:02:11] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: v=0 (3)
[Mar 10 12:02:11] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: o=root 15131 15131 IN IP4 24.123.23.170 (39)
[Mar 10 12:02:11] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: s=session (9)
[Mar 10 12:02:11] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: c=IN IP4 24.123.23.170 (22)
[Mar 10 12:02:11] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: t=0 0 (5)
[Mar 10 12:02:11] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: m=audio 14572 RTP/AVP 0 8 3 101 (31)
[Mar 10 12:02:11] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: a=rtpmap:0 PCMU/8000 (20)
[Mar 10 12:02:11] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: a=rtpmap:8 PCMA/8000 (20)
[Mar 10 12:02:11] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: a=rtpmap:3 GSM/8000 (19)
[Mar 10 12:02:11] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33)
[Mar 10 12:02:11] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: a=fmtp:101 0-16 (15)
[Mar 10 12:02:11] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: a=silenceSupp:off - - - - (25)

[Mar 10 12:02:11] 13 headers, 12 lines

[Mar 10 12:02:11] Reliably Transmitting (no NAT) to 192.168.1.85:5060:
INVITE sip:523@192.168.1.85:5060 SIP/2.0
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK202ed764
From: "551 551" <sip:551@24.123.23.170>;tag=as6fedd0f9
To: <sip:523@192.168.1.85:5060>
Contact: <sip:551@24.123.23.170>
Call-ID: 3890f17d116db7343bb188d42c093fd2@24.123.23.170
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 10 Mar 2007 17:02:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 15131 15131 IN IP4 24.123.23.170
s=session
c=IN IP4 24.123.23.170
t=0 0
m=audio 14572 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 - - - -

---
[Mar 10 12:02:11] DEBUG[15201]: chan_sip.c:1304 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id  #118

[Mar 10 12:02:11]     -- Called 523

[Mar 10 12:02:11] 
<-- SIP read from 192.168.1.85:5060: 
SIP/2.0 100 Trying
To: <sip:523@192.168.1.85:5060>;tag=a997c53df28f4a860026417db9e2f2ab
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK202ed764
From: "551 551" <sip:551@24.123.23.170>;tag=as6fedd0f9
Call-ID: 3890f17d116db7343bb188d42c093fd2@24.123.23.170
CSeq: 102 INVITE
Content-Length: 0
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK


[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: SIP/2.0 100 Trying (18)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: To: <sip:523@192.168.1.85:5060>;tag=a997c53df28f4a860026417db9e2f2ab (68)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK202ed764 (58)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: From: "551 551" <sip:551@24.123.23.170>;tag=as6fedd0f9 (54)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Call-ID: 3890f17d116db7343bb188d42c093fd2@24.123.23.170 (55)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: CSeq: 102 INVITE (16)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: Content-Length: 0 (17)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Supported: sip-cc, sip-cc-01, replaces, timer (45)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9:  (0)

[Mar 10 12:02:11] --- (9 headers 0 lines) ---
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:1456 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #118 - INVITE (got response)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:1465 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '3890f17d116db7343bb188d42c093fd2@24.123.23.170' Request 102: Found
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:9642 handle_response_invite: SIP response 100 to standard invite

[Mar 10 12:02:11] 
<-- SIP read from 192.168.1.85:5060: 
SIP/2.0 180 Ringing
To: <sip:523@192.168.1.85:5060>;tag=a997c53df28f4a860026417db9e2f2ab
Contact: <sip:523@192.168.1.85:5060>
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK202ed764
From: "551 551" <sip:551@24.123.23.170>;tag=as6fedd0f9
Call-ID: 3890f17d116db7343bb188d42c093fd2@24.123.23.170
CSeq: 102 INVITE
Content-Length: 0
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK


[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: SIP/2.0 180 Ringing (19)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: To: <sip:523@192.168.1.85:5060>;tag=a997c53df28f4a860026417db9e2f2ab (68)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: Contact: <sip:523@192.168.1.85:5060> (36)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK202ed764 (58)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: From: "551 551" <sip:551@24.123.23.170>;tag=as6fedd0f9 (54)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: Call-ID: 3890f17d116db7343bb188d42c093fd2@24.123.23.170 (55)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: CSeq: 102 INVITE (16)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Content-Length: 0 (17)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Supported: sip-cc, sip-cc-01, replaces, timer (45)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62)
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10:  (0)

[Mar 10 12:02:11] --- (10 headers 0 lines) ---
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:1465 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '3890f17d116db7343bb188d42c093fd2@24.123.23.170' Request 102: Found
[Mar 10 12:02:11] DEBUG[15149]: chan_sip.c:9642 handle_response_invite: SIP response 180 to standard invite

[Mar 10 12:02:11]     -- SIP/523-08521b58 is ringing

[Mar 10 12:02:11] Transmitting (NAT) to 192.168.1.97:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.97:5060;branch=z9hG4bK3702870459;received=192.168.1.97;rport=5060
From: "Display Name" <sip:551@24.123.23.170>;tag=1341134544
To: <sip:523@24.123.23.170>;tag=as57536955
Call-ID: 3175191617@192.168.1.97
CSeq: 1001 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:523@24.123.23.170>
Content-Length: 0


---
[Mar 10 12:02:11] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 523
[Mar 10 12:02:11] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/523 - state 6 (Ringing)
[Mar 10 12:02:11] DEBUG[15204]: app_queue.c:500 changethread: Device 'SIP/523' changed to state '6' (Ringing) but we don't care because they're not a member of any queue.

[Mar 10 12:02:11] 
<-- SIP read from 192.168.1.86:5060: 


[Mar 10 12:02:11] --- (0 headers 0 lines) Nat keepalive ---

[Mar 10 12:02:12] 
<-- SIP read from 192.168.1.85:5060: 
INVITE sip:593@24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKw4a30016f8fc6d9cc243d2adc53e4f190
Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170
CSeq: 388768 INVITE
From: 523 <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f
To: <sip:593@24.123.23.170>;tag=as0da2dd59
Contact: <sip:523@192.168.1.85:5060>
Content-Type: application/sdp
User-Agent: Uniden SIP Phone p2 Ver BS4.77
Max-Forwards: 70
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK
Content-Length: 218

v=0
o=- 15131 248057 IN IP4 192.168.1.85
s=session
c=IN IP4 0.0.0.0
t=0 0
m=audio 30012 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,16,17,18
a=silenceSupp:off - - - -

[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: INVITE sip:593@24.123.23.170 SIP/2.0 (36)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKw4a30016f8fc6d9cc243d2adc53e4f190 (82)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170 (55)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: CSeq: 388768 INVITE (19)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: From: 523 <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f (74)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: To: <sip:593@24.123.23.170>;tag=as0da2dd59 (42)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: Contact: <sip:523@192.168.1.85:5060> (36)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Content-Type: application/sdp (29)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Max-Forwards: 70 (16)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 12: Content-Length: 218 (19)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 13:  (0)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: v=0 (3)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: o=- 15131 248057 IN IP4 192.168.1.85 (36)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: s=session (9)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: c=IN IP4 0.0.0.0 (16)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: t=0 0 (5)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: m=audio 30012 RTP/AVP 0 101 (27)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:0 PCMU/8000 (20)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=fmtp:101 0-15,16,17,18 (24)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=silenceSupp:off - - - - (25)

[Mar 10 12:02:12] --- (13 headers 10 lines) ---
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received INVITE (5) - Command in SIP INVITE
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:1015 parse_sip_options: Begin: parsing SIP "Supported: sip-cc, sip-cc-01, replaces, timer"
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:1027 parse_sip_options: Found SIP option: -sip-cc-
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:1038 parse_sip_options: Found no match for SIP option: sip-cc (Please file bug report!)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:1027 parse_sip_options: Found SIP option: -sip-cc-01-
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:1038 parse_sip_options: Found no match for SIP option: sip-cc-01 (Please file bug report!)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:1027 parse_sip_options: Found SIP option: -replaces-
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:1033 parse_sip_options: Matched SIP option: replaces
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:1027 parse_sip_options: Found SIP option: -timer-
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:1033 parse_sip_options: Matched SIP option: timer
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:1044 parse_sip_options: * SIP extension value: 5 for call 5d28f46f54fd450475a9f02419f99674@24.123.23.170

[Mar 10 12:02:12] Using INVITE request as basis request - 5d28f46f54fd450475a9f02419f99674@24.123.23.170

[Mar 10 12:02:12] Sending to 192.168.1.85 : 5060 (non-NAT)

[Mar 10 12:02:12] Found RTP audio format 0

[Mar 10 12:02:12] Found RTP audio format 101

[Mar 10 12:02:12] Peer audio RTP is at port 0.0.0.0:30012
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3631 process_sdp: Peer audio RTP is at port 0.0.0.0:30012

[Mar 10 12:02:12] Peer video RTP is at port 0.0.0.0:65535
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3653 process_sdp: Peer video RTP is at port 0.0.0.0:65535

[Mar 10 12:02:12] Found description format PCMU

[Mar 10 12:02:12] Found description format telephone-event

[Mar 10 12:02:12] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)

[Mar 10 12:02:12] Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Mar 10 12:02:12] DEBUG[15149]: channel.c:2409 set_format: Set channel SIP/593-084f44b8 to write format slin

[Mar 10 12:02:12]     -- Started music on hold, class 'default', on channel 'SIP/593-084f44b8'
[Mar 10 12:02:12] DEBUG[15149]: channel.c:1761 ast_settimeout: Scheduling timer at 160 sample intervals
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:10625 handle_request_invite: Got a SIP re-invite for call 5d28f46f54fd450475a9f02419f99674@24.123.23.170

[Mar 10 12:02:12] We're at 24.123.23.170 port 17986

[Mar 10 12:02:12] Video is at 24.123.23.170 port 12600

[Mar 10 12:02:12] Adding codec 0x4 (ulaw) to SDP

[Mar 10 12:02:12] Adding codec 0x8 (alaw) to SDP

[Mar 10 12:02:12] Adding codec 0x100 (g729) to SDP

[Mar 10 12:02:12] Adding non-codec 0x1 (telephone-event) to SDP

[Mar 10 12:02:12] Reliably Transmitting (no NAT) to 192.168.1.85:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKw4a30016f8fc6d9cc243d2adc53e4f190;received=192.168.1.85
From: 523 <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f
To: <sip:593@24.123.23.170>;tag=as0da2dd59
Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170
CSeq: 388768 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:593@24.123.23.170>
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 15131 15133 IN IP4 192.168.1.83
s=session
c=IN IP4 192.168.1.83
t=0 0
m=audio 51000 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:1304 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id  #119
[Mar 10 12:02:12] DEBUG[15195]: rtp.c:1672 ast_rtp_bridge: Oooh, 'SIP/523-0850b038' changed end address to 0.0.0.0:30012 (format 4)
[Mar 10 12:02:12] DEBUG[15195]: rtp.c:1674 ast_rtp_bridge: Oooh, 'SIP/523-0850b038' changed end vaddress to 0.0.0.0:65535 (format 4)
[Mar 10 12:02:12] DEBUG[15195]: rtp.c:1676 ast_rtp_bridge: Oooh, 'SIP/523-0850b038' was 192.168.1.85:30012/(format 4)
[Mar 10 12:02:12] DEBUG[15195]: rtp.c:1678 ast_rtp_bridge: Oooh, 'SIP/523-0850b038' was 192.168.1.85:65535/(format 4)
[Mar 10 12:02:12] DEBUG[15195]: chan_sip.c:13028 sip_set_rtp_peer: Sending reinvite on SIP '20a99672af308312d36264481f6b7f5a@192.168.1.83' - It's audio soon redirected to IP 24.123.23.170

[Mar 10 12:02:12] set_destination: Parsing <sip:593@192.168.1.83:5060> for address/port to send to

[Mar 10 12:02:12] set_destination: set destination to 192.168.1.83, port 5060

[Mar 10 12:02:12] We're at 24.123.23.170 port 19424

[Mar 10 12:02:12] Video is at 24.123.23.170 port 13400

[Mar 10 12:02:12] Adding codec 0x4 (ulaw) to SDP

[Mar 10 12:02:12] Adding non-codec 0x1 (telephone-event) to SDP
[Mar 10 12:02:12] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 0: INVITE sip:593@192.168.1.83:5060 SIP/2.0 (40)
[Mar 10 12:02:12] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK1d52f739 (58)
[Mar 10 12:02:12] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 2: From: 523 <sip:523@24.123.23.170:5060>;tag=as38239579 (53)
[Mar 10 12:02:12] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 3: To: 593 <sip:593@24.123.23.170:5060>;tag=b3eeb4c9225baf5 (56)
[Mar 10 12:02:12] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 4: Contact: <sip:523@24.123.23.170> (32)
[Mar 10 12:02:12] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 5: Call-ID: 20a99672af308312d36264481f6b7f5a@192.168.1.83 (54)
[Mar 10 12:02:12] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 6: CSeq: 103 INVITE (16)
[Mar 10 12:02:12] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 7: User-Agent: Asterisk PBX (24)
[Mar 10 12:02:12] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 8: Max-Forwards: 70 (16)
[Mar 10 12:02:12] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66)
[Mar 10 12:02:12] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 10: X-asterisk-info: SIP re-invite (RTP bridge) (43)
[Mar 10 12:02:12] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 11: Content-Type: application/sdp (29)
[Mar 10 12:02:12] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 12: Content-Length: 218 (19)
[Mar 10 12:02:12] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 13:  (0)
[Mar 10 12:02:12] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: v=0 (3)
[Mar 10 12:02:12] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: o=root 15131 15133 IN IP4 24.123.23.170 (39)
[Mar 10 12:02:12] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: s=session (9)
[Mar 10 12:02:12] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: c=IN IP4 24.123.23.170 (22)
[Mar 10 12:02:12] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: t=0 0 (5)
[Mar 10 12:02:12] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: m=audio 19424 RTP/AVP 0 101 (27)
[Mar 10 12:02:12] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: a=rtpmap:0 PCMU/8000 (20)
[Mar 10 12:02:12] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33)
[Mar 10 12:02:12] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: a=fmtp:101 0-16 (15)
[Mar 10 12:02:12] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: a=silenceSupp:off - - - - (25)

[Mar 10 12:02:12] 13 headers, 10 lines

[Mar 10 12:02:12] Reliably Transmitting (no NAT) to 192.168.1.83:5060:
INVITE sip:593@192.168.1.83:5060 SIP/2.0
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK1d52f739
From: 523 <sip:523@24.123.23.170:5060>;tag=as38239579
To: 593 <sip:593@24.123.23.170:5060>;tag=b3eeb4c9225baf5
Contact: <sip:523@24.123.23.170>
Call-ID: 20a99672af308312d36264481f6b7f5a@192.168.1.83
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 218

v=0
o=root 15131 15133 IN IP4 24.123.23.170
s=session
c=IN IP4 24.123.23.170
t=0 0
m=audio 19424 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
[Mar 10 12:02:12] DEBUG[15195]: chan_sip.c:1304 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id  #120
[Mar 10 12:02:12] DEBUG[15195]: rtp.c:1359 ast_rtp_write: Ooh, format changed from unknown to ulaw

[Mar 10 12:02:12] 
<-- SIP read from 192.168.1.85:5060: 
ACK sip:593@24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKmb09c6ae76bb5fb0eb40a6ee81e316b8b
CSeq: 388768 ACK
To: <sip:593@24.123.23.170>;tag=as0da2dd59
Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170
From: 523 <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f
User-Agent: Uniden SIP Phone p2 Ver BS4.77


[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: ACK sip:593@24.123.23.170 SIP/2.0 (33)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKmb09c6ae76bb5fb0eb40a6ee81e316b8b (82)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: CSeq: 388768 ACK (16)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: To: <sip:593@24.123.23.170>;tag=as0da2dd59 (42)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170 (55)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: From: 523 <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f (74)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7:  (0)

[Mar 10 12:02:12] --- (7 headers 0 lines) ---
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received ACK (6) - Command in SIP ACK
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:1401 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #119
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:1412 __sip_ack: Stopping retransmission on '5d28f46f54fd450475a9f02419f99674@24.123.23.170' of Response 388768: Match Found

[Mar 10 12:02:12] 
<-- SIP read from 192.168.1.83:5060: 
SIP/2.0 200 OK
Call-ID: 20a99672af308312d36264481f6b7f5a@192.168.1.83
CSeq: 103 INVITE
From: 523 <sip:523@24.123.23.170:5060>;tag=as38239579
To: 593 <sip:593@24.123.23.170:5060>;tag=b3eeb4c9225baf5
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK1d52f739
Content-Length: 230
Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE
Content-Type: application/sdp
Supported: replaces
Contact: 593 <sip:593@192.168.1.83:5060>
User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4

v=0
o=MxSIP 0 1158539950 IN IP4 192.168.1.83
s=SIP Call
c=IN IP4 192.168.1.83
t=0 0
m=audio 51000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:on - - - -

[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: SIP/2.0 200 OK (14)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Call-ID: 20a99672af308312d36264481f6b7f5a@192.168.1.83 (54)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: CSeq: 103 INVITE (16)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: From: 523 <sip:523@24.123.23.170:5060>;tag=as38239579 (53)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: To: 593 <sip:593@24.123.23.170:5060>;tag=b3eeb4c9225baf5 (56)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK1d52f739 (58)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: Content-Length: 230 (19)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE (55)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Content-Type: application/sdp (29)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Supported: replaces (19)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Contact: 593 <sip:593@192.168.1.83:5060> (40)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11: User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 (42)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 12:  (0)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: v=0 (3)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: o=MxSIP 0 1158539950 IN IP4 192.168.1.83 (40)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: s=SIP Call (10)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: c=IN IP4 192.168.1.83 (21)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: t=0 0 (5)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: m=audio 51000 RTP/AVP 0 101 (27)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:0 PCMU/8000 (20)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=fmtp:101 0-15 (15)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=ptime:20 (10)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=silenceSupp:on - - - - (24)

[Mar 10 12:02:12] --- (12 headers 11 lines) ---
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:1390 __sip_ack: Acked pending invite 103
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:1401 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #120
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:1412 __sip_ack: Stopping retransmission on '20a99672af308312d36264481f6b7f5a@192.168.1.83' of Request 103: Match Found
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:9640 handle_response_invite: SIP response 200 to RE-invite on outgoing call 20a99672af308312d36264481f6b7f5a@192.168.1.83

[Mar 10 12:02:12] Found RTP audio format 0

[Mar 10 12:02:12] Found RTP audio format 101

[Mar 10 12:02:12] Peer audio RTP is at port 192.168.1.83:51000
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3631 process_sdp: Peer audio RTP is at port 192.168.1.83:51000

[Mar 10 12:02:12] Peer video RTP is at port 192.168.1.83:65535
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3653 process_sdp: Peer video RTP is at port 192.168.1.83:65535

[Mar 10 12:02:12] Found description format PCMU

[Mar 10 12:02:12] Found description format telephone-event

[Mar 10 12:02:12] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)

[Mar 10 12:02:12] Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:6137 build_route: build_route: Retaining previous route: <sip:593@192.168.1.83:5060>

[Mar 10 12:02:12] set_destination: Parsing <sip:593@192.168.1.83:5060> for address/port to send to

[Mar 10 12:02:12] set_destination: set destination to 192.168.1.83, port 5060

[Mar 10 12:02:12] Transmitting (no NAT) to 192.168.1.83:5060:
ACK sip:593@192.168.1.83:5060 SIP/2.0
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK39d1677d
From: 523 <sip:523@24.123.23.170:5060>;tag=as38239579
To: 593 <sip:593@24.123.23.170:5060>;tag=b3eeb4c9225baf5
Contact: <sip:523@24.123.23.170>
Call-ID: 20a99672af308312d36264481f6b7f5a@192.168.1.83
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---

[Mar 10 12:02:12] 
<-- SIP read from 192.168.1.85:5060: 
SIP/2.0 200 OK
To: <sip:523@192.168.1.85:5060>;tag=a997c53df28f4a860026417db9e2f2ab
Contact: <sip:523@192.168.1.85:5060>
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK202ed764
From: "551 551" <sip:551@24.123.23.170>;tag=as6fedd0f9
Call-ID: 3890f17d116db7343bb188d42c093fd2@24.123.23.170
CSeq: 102 INVITE
Content-Type: application/sdp
Content-Length: 223
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK

v=0
o=- 15131 298887 IN IP4 192.168.1.85
s=session
c=IN IP4 192.168.1.85
t=0 0
m=audio 30014 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,16,17,18
a=silenceSupp:off - - - -

[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: SIP/2.0 200 OK (14)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: To: <sip:523@192.168.1.85:5060>;tag=a997c53df28f4a860026417db9e2f2ab (68)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: Contact: <sip:523@192.168.1.85:5060> (36)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK202ed764 (58)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: From: "551 551" <sip:551@24.123.23.170>;tag=as6fedd0f9 (54)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: Call-ID: 3890f17d116db7343bb188d42c093fd2@24.123.23.170 (55)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: CSeq: 102 INVITE (16)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Content-Type: application/sdp (29)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Content-Length: 223 (19)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Supported: sip-cc, sip-cc-01, replaces, timer (45)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11:  (0)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: v=0 (3)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: o=- 15131 298887 IN IP4 192.168.1.85 (36)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: s=session (9)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: c=IN IP4 192.168.1.85 (21)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: t=0 0 (5)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: m=audio 30014 RTP/AVP 0 101 (27)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:0 PCMU/8000 (20)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=fmtp:101 0-15,16,17,18 (24)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=silenceSupp:off - - - - (25)

[Mar 10 12:02:12] --- (11 headers 10 lines) ---
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:1390 __sip_ack: Acked pending invite 102
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:1412 __sip_ack: Stopping retransmission on '3890f17d116db7343bb188d42c093fd2@24.123.23.170' of Request 102: Match Found
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:9642 handle_response_invite: SIP response 200 to standard invite

[Mar 10 12:02:12] Found RTP audio format 0

[Mar 10 12:02:12] Found RTP audio format 101

[Mar 10 12:02:12] Peer audio RTP is at port 192.168.1.85:30014
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3631 process_sdp: Peer audio RTP is at port 192.168.1.85:30014

[Mar 10 12:02:12] Peer video RTP is at port 192.168.1.85:65535
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:3653 process_sdp: Peer video RTP is at port 192.168.1.85:65535

[Mar 10 12:02:12] Found description format PCMU

[Mar 10 12:02:12] Found description format telephone-event

[Mar 10 12:02:12] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)

[Mar 10 12:02:12] Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Mar 10 12:02:12] DEBUG[15149]: chan_sip.c:6194 build_route: build_route: Contact hop: <sip:523@192.168.1.85:5060>

[Mar 10 12:02:12] list_route: hop: <sip:523@192.168.1.85:5060>

[Mar 10 12:02:12] set_destination: Parsing <sip:523@192.168.1.85:5060> for address/port to send to

[Mar 10 12:02:12] set_destination: set destination to 192.168.1.85, port 5060

[Mar 10 12:02:12] Transmitting (no NAT) to 192.168.1.85:5060:
ACK sip:523@192.168.1.85:5060 SIP/2.0
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK508927ba
From: "551 551" <sip:551@24.123.23.170>;tag=as6fedd0f9
To: <sip:523@192.168.1.85:5060>;tag=a997c53df28f4a860026417db9e2f2ab
Contact: <sip:551@24.123.23.170>
Call-ID: 3890f17d116db7343bb188d42c093fd2@24.123.23.170
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
[Mar 10 12:02:12] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 523
[Mar 10 12:02:12] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/523 - state 2 (In use)
[Mar 10 12:02:12] DEBUG[15205]: app_queue.c:500 changethread: Device 'SIP/523' changed to state '2' (In use) but we don't care because they're not a member of any queue.

[Mar 10 12:02:12]     -- SIP/523-08521b58 answered SIP/551-08518cd8
[Mar 10 12:02:12] DEBUG[15201]: chan_sip.c:2557 sip_answer: sip_answer(SIP/551-08518cd8)

[Mar 10 12:02:12] We're at 24.123.23.170 port 19120

[Mar 10 12:02:12] Video is at 24.123.23.170 port 13200

[Mar 10 12:02:12] Adding codec 0x4 (ulaw) to SDP
[Mar 10 12:02:12] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 551

[Mar 10 12:02:12] DEBUG[15136]: channel.c:777 channel_find_locked: Avoiding initial deadlock for 'SIP/551-08518cd8'
[Mar 10 12:02:12] Adding codec 0x8 (alaw) to SDP

[Mar 10 12:02:12] Adding non-codec 0x1 (telephone-event) to SDP

[Mar 10 12:02:12] Reliably Transmitting (NAT) to 192.168.1.97:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.97:5060;branch=z9hG4bK3702870459;received=192.168.1.97;rport=5060
From: "Display Name" <sip:551@24.123.23.170>;tag=1341134544
To: <sip:523@24.123.23.170>;tag=as57536955
Call-ID: 3175191617@192.168.1.97
CSeq: 1001 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:523@24.123.23.170>
Content-Type: application/sdp
Content-Length: 242

v=0
o=root 15131 15131 IN IP4 24.123.23.170
s=session
c=IN IP4 24.123.23.170
t=0 0
m=audio 19120 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
[Mar 10 12:02:12] DEBUG[15201]: chan_sip.c:1304 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id  #121

[Mar 10 12:02:12]     -- Attempting native bridge of SIP/551-08518cd8 and SIP/523-08521b58
[Mar 10 12:02:12] DEBUG[15201]: chan_sip.c:13034 sip_set_rtp_peer: Deferring reinvite on SIP '3175191617@192.168.1.97' - It's audio will be redirected to IP 192.168.1.85
[Mar 10 12:02:12] DEBUG[15201]: chan_sip.c:13028 sip_set_rtp_peer: Sending reinvite on SIP '3890f17d116db7343bb188d42c093fd2@24.123.23.170' - It's audio soon redirected to IP 192.168.1.97

[Mar 10 12:02:12] set_destination: Parsing <sip:523@192.168.1.85:5060> for address/port to send to

[Mar 10 12:02:12] set_destination: set destination to 192.168.1.85, port 5060

[Mar 10 12:02:12] We're at 24.123.23.170 port 14572

[Mar 10 12:02:12] Video is at 24.123.23.170 port 16640

[Mar 10 12:02:12] Adding codec 0x4 (ulaw) to SDP

[Mar 10 12:02:12] Adding codec 0x8 (alaw) to SDP

[Mar 10 12:02:12] Adding codec 0x10 (g726) to SDP

[Mar 10 12:02:12] Adding codec 0x100 (g729) to SDP

[Mar 10 12:02:12] Adding non-codec 0x1 (telephone-event) to SDP
[Mar 10 12:02:12] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 0: INVITE sip:523@192.168.1.85:5060 SIP/2.0 (40)
[Mar 10 12:02:12] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK3cbfffca (58)
[Mar 10 12:02:12] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 2: From: "551 551" <sip:551@24.123.23.170>;tag=as6fedd0f9 (54)
[Mar 10 12:02:12] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 3: To: <sip:523@192.168.1.85:5060>;tag=a997c53df28f4a860026417db9e2f2ab (68)
[Mar 10 12:02:12] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 4: Contact: <sip:551@24.123.23.170> (32)
[Mar 10 12:02:12] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 5: Call-ID: 3890f17d116db7343bb188d42c093fd2@24.123.23.170 (55)
[Mar 10 12:02:12] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 6: CSeq: 103 INVITE (16)
[Mar 10 12:02:12] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 7: User-Agent: Asterisk PBX (24)
[Mar 10 12:02:12] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 8: Max-Forwards: 70 (16)
[Mar 10 12:02:12] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66)
[Mar 10 12:02:12] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 10: X-asterisk-info: SIP re-invite (RTP bridge) (43)
[Mar 10 12:02:12] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 11: Content-Type: application/sdp (29)
[Mar 10 12:02:12] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 12: Content-Length: 318 (19)
[Mar 10 12:02:12] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 13:  (0)
[Mar 10 12:02:12] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: v=0 (3)
[Mar 10 12:02:12] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: o=root 15131 15132 IN IP4 192.168.1.97 (38)
[Mar 10 12:02:12] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: s=session (9)
[Mar 10 12:02:12] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: c=IN IP4 192.168.1.97 (21)
[Mar 10 12:02:12] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: t=0 0 (5)
[Mar 10 12:02:12] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: m=audio 10010 RTP/AVP 0 8 111 18 101 (36)
[Mar 10 12:02:12] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: a=rtpmap:0 PCMU/8000 (20)
[Mar 10 12:02:12] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: a=rtpmap:8 PCMA/8000 (20)
[Mar 10 12:02:12] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: a=rtpmap:111 G726-32/8000 (25)
[Mar 10 12:02:12] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: a=rtpmap:18 G729/8000 (21)
[Mar 10 12:02:12] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: a=fmtp:18 annexb=no (19)
[Mar 10 12:02:12] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33)
[Mar 10 12:02:12] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: a=fmtp:101 0-16 (15)
[Mar 10 12:02:12] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: a=silenceSupp:off - - - - (25)

[Mar 10 12:02:12] 13 headers, 14 lines

[Mar 10 12:02:12] Reliably Transmitting (no NAT) to 192.168.1.85:5060:
INVITE sip:523@192.168.1.85:5060 SIP/2.0
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK3cbfffca
From: "551 551" <sip:551@24.123.23.170>;tag=as6fedd0f9
To: <sip:523@192.168.1.85:5060>;tag=a997c53df28f4a860026417db9e2f2ab
Contact: <sip:551@24.123.23.170>
Call-ID: 3890f17d116db7343bb188d42c093fd2@24.123.23.170
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 318

v=0
o=root 15131 15132 IN IP4 192.168.1.97
s=session
c=IN IP4 192.168.1.97
t=0 0
m=audio 10010 RTP/AVP 0 8 111 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
[Mar 10 12:02:12] DEBUG[15201]: chan_sip.c:1304 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id  #122
[Mar 10 12:02:12] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/551 - state 2 (In use)
[Mar 10 12:02:12] DEBUG[15206]: app_queue.c:500 changethread: Device 'SIP/551' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[Mar 10 12:02:13] DEBUG[15195]: channel.c:2027 ast_read: Generator got voice, switching to phase locked mode
[Mar 10 12:02:13] DEBUG[15195]: channel.c:1761 ast_settimeout: Scheduling timer at 0 sample intervals
[Mar 10 12:02:13] DEBUG[15201]: rtp.c:411 ast_rtcp_read: Got RTCP report of 64 bytes
[Mar 10 12:02:13] DEBUG[15201]: rtp.c:1359 ast_rtp_write: Ooh, format changed from unknown to ulaw

[Mar 10 12:02:13] 
<-- SIP read from 192.168.1.97:5060: 
ACK sip:523@24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.97:5060;rport;branch=z9hG4bK2925421636
From: "Display Name" <sip:551@24.123.23.170>;tag=1341134544
To: <sip:523@24.123.23.170>;tag=as57536955
Call-ID: 3175191617@192.168.1.97
CSeq: 1001 ACK
max-forwards: 70
user-agent: UTSTARCOM F1000/Device ID-0007ba26174b
Content-Length: 0


[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: ACK sip:523@24.123.23.170 SIP/2.0 (33)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.97:5060;rport;branch=z9hG4bK2925421636 (65)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: From: "Display Name" <sip:551@24.123.23.170>;tag=1341134544 (59)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: To: <sip:523@24.123.23.170>;tag=as57536955 (42)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Call-ID: 3175191617@192.168.1.97 (32)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: CSeq: 1001 ACK (14)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: max-forwards: 70 (16)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: user-agent: UTSTARCOM F1000/Device ID-0007ba26174b (50)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Content-Length: 0 (17)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9:  (0)

[Mar 10 12:02:13] --- (9 headers 0 lines) ---
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received ACK (6) - Command in SIP ACK
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:1401 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #121
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:1412 __sip_ack: Stopping retransmission on '3175191617@192.168.1.97' of Response 1001: Match Found
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:9625 check_pendings: Sending pending reinvite on '3175191617@192.168.1.97'

[Mar 10 12:02:13] set_destination: Parsing <sip:551@192.168.1.97:5060> for address/port to send to

[Mar 10 12:02:13] set_destination: set destination to 192.168.1.97, port 5060

[Mar 10 12:02:13] We're at 24.123.23.170 port 19120

[Mar 10 12:02:13] Video is at 24.123.23.170 port 13200

[Mar 10 12:02:13] Adding codec 0x4 (ulaw) to SDP

[Mar 10 12:02:13] Adding non-codec 0x1 (telephone-event) to SDP
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: INVITE sip:551@192.168.1.97:5060 SIP/2.0 (40)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK1d3f1606;rport (64)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: From: <sip:523@24.123.23.170>;tag=as57536955 (44)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: To: "Display Name" <sip:551@24.123.23.170>;tag=1341134544 (57)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Contact: <sip:523@24.123.23.170> (32)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: Call-ID: 3175191617@192.168.1.97 (32)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: CSeq: 102 INVITE (16)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: User-Agent: Asterisk PBX (24)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Max-Forwards: 70 (16)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: X-asterisk-info: SIP re-invite (RTP bridge) (43)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11: Content-Type: application/sdp (29)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 12: Content-Length: 216 (19)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 13:  (0)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: v=0 (3)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: o=root 15131 15132 IN IP4 192.168.1.85 (38)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: s=session (9)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: c=IN IP4 192.168.1.85 (21)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: t=0 0 (5)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: m=audio 30014 RTP/AVP 0 101 (27)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:0 PCMU/8000 (20)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=fmtp:101 0-16 (15)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=silenceSupp:off - - - - (25)

[Mar 10 12:02:13] 13 headers, 10 lines

[Mar 10 12:02:13] Reliably Transmitting (NAT) to 192.168.1.97:5060:
INVITE sip:551@192.168.1.97:5060 SIP/2.0
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK1d3f1606;rport
From: <sip:523@24.123.23.170>;tag=as57536955
To: "Display Name" <sip:551@24.123.23.170>;tag=1341134544
Contact: <sip:523@24.123.23.170>
Call-ID: 3175191617@192.168.1.97
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 216

v=0
o=root 15131 15132 IN IP4 192.168.1.85
s=session
c=IN IP4 192.168.1.85
t=0 0
m=audio 30014 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:1304 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id  #123

[Mar 10 12:02:13] 
<-- SIP read from 192.168.1.85:5060: 
SIP/2.0 200 OK
To: <sip:523@192.168.1.85:5060>;tag=a997c53df28f4a860026417db9e2f2ab
Contact: <sip:523@192.168.1.85:5060>
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK3cbfffca
From: "551 551" <sip:551@24.123.23.170>;tag=as6fedd0f9
Call-ID: 3890f17d116db7343bb188d42c093fd2@24.123.23.170
CSeq: 103 INVITE
Content-Type: application/sdp
Content-Length: 197
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK

v=0
o=- 15131 298888 IN IP4 192.168.1.85
s=session
c=IN IP4 192.168.1.85
t=0 0
m=audio 30014 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -

[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: SIP/2.0 200 OK (14)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: To: <sip:523@192.168.1.85:5060>;tag=a997c53df28f4a860026417db9e2f2ab (68)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: Contact: <sip:523@192.168.1.85:5060> (36)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK3cbfffca (58)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: From: "551 551" <sip:551@24.123.23.170>;tag=as6fedd0f9 (54)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: Call-ID: 3890f17d116db7343bb188d42c093fd2@24.123.23.170 (55)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: CSeq: 103 INVITE (16)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Content-Type: application/sdp (29)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Content-Length: 197 (19)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Supported: sip-cc, sip-cc-01, replaces, timer (45)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11:  (0)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: v=0 (3)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: o=- 15131 298888 IN IP4 192.168.1.85 (36)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: s=session (9)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: c=IN IP4 192.168.1.85 (21)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: t=0 0 (5)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: m=audio 30014 RTP/AVP 0 101 (27)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:0 PCMU/8000 (20)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=silenceSupp:off - - - - (25)

[Mar 10 12:02:13] --- (11 headers 9 lines) ---
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:1390 __sip_ack: Acked pending invite 103
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:1401 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #122
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:1412 __sip_ack: Stopping retransmission on '3890f17d116db7343bb188d42c093fd2@24.123.23.170' of Request 103: Match Found
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:9640 handle_response_invite: SIP response 200 to RE-invite on outgoing call 3890f17d116db7343bb188d42c093fd2@24.123.23.170

[Mar 10 12:02:13] Found RTP audio format 0

[Mar 10 12:02:13] Found RTP audio format 101

[Mar 10 12:02:13] Peer audio RTP is at port 192.168.1.85:30014
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3631 process_sdp: Peer audio RTP is at port 192.168.1.85:30014

[Mar 10 12:02:13] Peer video RTP is at port 192.168.1.85:65535
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3653 process_sdp: Peer video RTP is at port 192.168.1.85:65535

[Mar 10 12:02:13] Found description format PCMU

[Mar 10 12:02:13] Found description format telephone-event

[Mar 10 12:02:13] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)

[Mar 10 12:02:13] Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:6137 build_route: build_route: Retaining previous route: <sip:523@192.168.1.85:5060>

[Mar 10 12:02:13] set_destination: Parsing <sip:523@192.168.1.85:5060> for address/port to send to

[Mar 10 12:02:13] set_destination: set destination to 192.168.1.85, port 5060

[Mar 10 12:02:13] Transmitting (no NAT) to 192.168.1.85:5060:
ACK sip:523@192.168.1.85:5060 SIP/2.0
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK067f2565
From: "551 551" <sip:551@24.123.23.170>;tag=as6fedd0f9
To: <sip:523@192.168.1.85:5060>;tag=a997c53df28f4a860026417db9e2f2ab
Contact: <sip:551@24.123.23.170>
Call-ID: 3890f17d116db7343bb188d42c093fd2@24.123.23.170
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---

[Mar 10 12:02:13] 
<-- SIP read from 192.168.1.97:5060: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK1d3f1606;rport=5060
From: <sip:523@24.123.23.170>;tag=as57536955
To: "Display Name" <sip:551@24.123.23.170>;tag=1341134544
Call-ID: 3175191617@192.168.1.97
CSeq: 102 INVITE
Contact: <sip:551@192.168.1.97:5060>
Content-Type: application/sdp
Content-Length: 199  

v=0
o=551 123456 654330 IN IP4 192.168.1.97
s=none
c=IN IP4 192.168.1.97
t=0 0
m=audio 10010 RTP/AVP 0 101
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: SIP/2.0 200 OK (14)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK1d3f1606;rport=5060 (69)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: From: <sip:523@24.123.23.170>;tag=as57536955 (44)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: To: "Display Name" <sip:551@24.123.23.170>;tag=1341134544 (57)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Call-ID: 3175191617@192.168.1.97 (32)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: CSeq: 102 INVITE (16)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: Contact: <sip:551@192.168.1.97:5060> (36)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Content-Type: application/sdp (29)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Content-Length: 199   (21)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9:  (0)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: v=0 (3)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: o=551 123456 654330 IN IP4 192.168.1.97 (39)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: s=none (6)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: c=IN IP4 192.168.1.97 (21)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: t=0 0 (5)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: m=audio 10010 RTP/AVP 0 101 (27)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=ptime:20 (10)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:0 PCMU/8000 (20)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=fmtp:101 0-16 (15)

[Mar 10 12:02:13] --- (9 headers 10 lines) ---
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:1390 __sip_ack: Acked pending invite 102
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:1401 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #123
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:1412 __sip_ack: Stopping retransmission on '3175191617@192.168.1.97' of Request 102: Match Found
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:9640 handle_response_invite: SIP response 200 to RE-invite on outgoing call 3175191617@192.168.1.97

[Mar 10 12:02:13] Found RTP audio format 0

[Mar 10 12:02:13] Found RTP audio format 101

[Mar 10 12:02:13] Peer audio RTP is at port 192.168.1.97:10010
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3631 process_sdp: Peer audio RTP is at port 192.168.1.97:10010

[Mar 10 12:02:13] Peer video RTP is at port 192.168.1.97:65535
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:3653 process_sdp: Peer video RTP is at port 192.168.1.97:65535

[Mar 10 12:02:13] Found description format PCMU

[Mar 10 12:02:13] Found description format telephone-event

[Mar 10 12:02:13] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)

[Mar 10 12:02:13] Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Mar 10 12:02:13] DEBUG[15149]: chan_sip.c:6194 build_route: build_route: Contact hop: <sip:551@192.168.1.97:5060>

[Mar 10 12:02:13] list_route: hop: <sip:551@192.168.1.97:5060>

[Mar 10 12:02:13] set_destination: Parsing <sip:551@192.168.1.97:5060> for address/port to send to

[Mar 10 12:02:13] set_destination: set destination to 192.168.1.97, port 5060

[Mar 10 12:02:13] Transmitting (NAT) to 192.168.1.97:5060:
ACK sip:551@192.168.1.97:5060 SIP/2.0
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5e2d313c;rport
From: <sip:523@24.123.23.170>;tag=as57536955
To: "Display Name" <sip:551@24.123.23.170>;tag=1341134544
Contact: <sip:523@24.123.23.170>
Call-ID: 3175191617@192.168.1.97
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
[Mar 10 12:02:13] DEBUG[15201]: rtp.c:411 ast_rtcp_read: Got RTCP report of 40 bytes
[Mar 10 12:02:14] DEBUG[15149]: chan_sip.c:3168 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP)
[Mar 10 12:02:14] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: OPTIONS sip:sip.broadvoice.com SIP/2.0 (38)
[Mar 10 12:02:14] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK680a5867;rport (64)
[Mar 10 12:02:14] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: From: "asterisk" <sip:asterisk@24.123.23.170>;tag=as107f44b5 (60)
[Mar 10 12:02:14] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: To: <sip:sip.broadvoice.com> (28)
[Mar 10 12:02:14] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Contact: <sip:asterisk@24.123.23.170> (37)
[Mar 10 12:02:14] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: Call-ID: 6d9e1d422128444604f53f4a39a0a95b@24.123.23.170 (55)
[Mar 10 12:02:14] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: CSeq: 102 OPTIONS (17)
[Mar 10 12:02:14] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: User-Agent: Asterisk PBX (24)
[Mar 10 12:02:14] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Max-Forwards: 70 (16)
[Mar 10 12:02:14] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Date: Sat, 10 Mar 2007 17:02:14 GMT (35)
[Mar 10 12:02:14] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66)
[Mar 10 12:02:14] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11: Content-Length: 0 (17)
[Mar 10 12:02:14] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 12:  (0)

[Mar 10 12:02:14] 12 headers, 0 lines

[Mar 10 12:02:14] Reliably Transmitting (no NAT) to 147.135.12.128:5060:
OPTIONS sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK680a5867;rport
From: "asterisk" <sip:asterisk@24.123.23.170>;tag=as107f44b5
To: <sip:sip.broadvoice.com>
Contact: <sip:asterisk@24.123.23.170>
Call-ID: 6d9e1d422128444604f53f4a39a0a95b@24.123.23.170
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 10 Mar 2007 17:02:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
[Mar 10 12:02:14] DEBUG[15149]: chan_sip.c:1304 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id  #124

[Mar 10 12:02:14] 
<-- SIP read from 147.135.12.128:5060: 
SIP/2.0 200 OK
Call-ID: 6d9e1d422128444604f53f4a39a0a95b@24.123.23.170
CSeq: 102 OPTIONS
From: "asterisk" <sip:asterisk@24.123.23.170>;tag=as107f44b5
To: <sip:sip.broadvoice.com>
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK680a5867
Supported: 100rel
Allow: INVITE, BYE, ACK, OPTIONS, CANCEL, PRACK
Accept: application/sdp
Accept-Encoding: 
Accept-Language: en
Content-Length:    0


[Mar 10 12:02:14] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: SIP/2.0 200 OK (14)
[Mar 10 12:02:14] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Call-ID: 6d9e1d422128444604f53f4a39a0a95b@24.123.23.170 (55)
[Mar 10 12:02:14] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: CSeq: 102 OPTIONS (17)
[Mar 10 12:02:14] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: From: "asterisk" <sip:asterisk@24.123.23.170>;tag=as107f44b5 (60)
[Mar 10 12:02:14] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: To: <sip:sip.broadvoice.com> (28)
[Mar 10 12:02:14] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK680a5867 (58)
[Mar 10 12:02:14] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: Supported: 100rel (17)
[Mar 10 12:02:14] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Allow: INVITE, BYE, ACK, OPTIONS, CANCEL, PRACK (47)
[Mar 10 12:02:14] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Accept: application/sdp (23)
[Mar 10 12:02:14] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Accept-Encoding:  (17)
[Mar 10 12:02:14] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Accept-Language: en (19)
[Mar 10 12:02:14] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11: Content-Length:    0 (20)
[Mar 10 12:02:14] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 12:  (0)

[Mar 10 12:02:14] --- (12 headers 0 lines) ---
[Mar 10 12:02:14] DEBUG[15149]: chan_sip.c:1401 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #124
[Mar 10 12:02:14] DEBUG[15149]: chan_sip.c:1412 __sip_ack: Stopping retransmission on '6d9e1d422128444604f53f4a39a0a95b@24.123.23.170' of Request 102: Match Found

[Mar 10 12:02:14] Destroying call '6d9e1d422128444604f53f4a39a0a95b@24.123.23.170'

[Mar 10 12:02:15] 
<-- SIP read from 192.168.1.97:5060: 
REGISTER sip:24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.97:5060;rport;branch=z9hG4bK3278849582
From: "Display Name" <sip:551@24.123.23.170>;tag=4282629717
To: "Display Name" <sip:551@24.123.23.170>
Call-ID: 1319586744@192.168.1.97
CSeq: 52 REGISTER
Contact: <sip:551@192.168.1.97:5060>;action=proxy
max-forwards: 70
expires: 60
user-agent: UTSTARCOM F1000/Device ID-0007ba26174b
Content-Length: 0


[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34)
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.97:5060;rport;branch=z9hG4bK3278849582 (65)
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: From: "Display Name" <sip:551@24.123.23.170>;tag=4282629717 (59)
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: To: "Display Name" <sip:551@24.123.23.170> (42)
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Call-ID: 1319586744@192.168.1.97 (32)
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: CSeq: 52 REGISTER (17)
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: Contact: <sip:551@192.168.1.97:5060>;action=proxy (49)
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: max-forwards: 70 (16)
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: expires: 60 (11)
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: user-agent: UTSTARCOM F1000/Device ID-0007ba26174b (50)
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Content-Length: 0 (17)
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11:  (0)

[Mar 10 12:02:15] --- (11 headers 0 lines) ---
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:3168 sip_alloc: Allocating new SIP dialog for 1319586744@192.168.1.97 - REGISTER (No RTP)
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:02:15] Using latest REGISTER request as basis request

[Mar 10 12:02:15] Sending to 192.168.1.97 : 5060 (NAT)

[Mar 10 12:02:15] Transmitting (NAT) to 192.168.1.97:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.97:5060;branch=z9hG4bK3278849582;received=192.168.1.97;rport=5060
From: "Display Name" <sip:551@24.123.23.170>;tag=4282629717
To: "Display Name" <sip:551@24.123.23.170>
Call-ID: 1319586744@192.168.1.97
CSeq: 52 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:551@24.123.23.170>
Content-Length: 0


---

[Mar 10 12:02:15] Transmitting (NAT) to 192.168.1.97:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.97:5060;branch=z9hG4bK3278849582;received=192.168.1.97;rport=5060
From: "Display Name" <sip:551@24.123.23.170>;tag=4282629717
To: "Display Name" <sip:551@24.123.23.170>;tag=as2e7b0a8c
Call-ID: 1319586744@192.168.1.97
CSeq: 52 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="30602214"
Content-Length: 0


---

[Mar 10 12:02:15] Scheduling destruction of call '1319586744@192.168.1.97' in 15000 ms

[Mar 10 12:02:15] 
<-- SIP read from 192.168.1.97:5060: 
REGISTER sip:24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.97:5060;rport;branch=z9hG4bK3278849582
From: "Display Name" <sip:551@24.123.23.170>;tag=4282629717
To: "Display Name" <sip:551@24.123.23.170>
Call-ID: 1319586744@192.168.1.97
CSeq: 52 REGISTER
Contact: <sip:551@192.168.1.97:5060>;action=proxy
max-forwards: 70
expires: 60
user-agent: UTSTARCOM F1000/Device ID-0007ba26174b
Content-Length: 0


[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34)
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.97:5060;rport;branch=z9hG4bK3278849582 (65)
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: From: "Display Name" <sip:551@24.123.23.170>;tag=4282629717 (59)
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: To: "Display Name" <sip:551@24.123.23.170> (42)
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Call-ID: 1319586744@192.168.1.97 (32)
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: CSeq: 52 REGISTER (17)
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: Contact: <sip:551@192.168.1.97:5060>;action=proxy (49)
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: max-forwards: 70 (16)
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: expires: 60 (11)
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: user-agent: UTSTARCOM F1000/Device ID-0007ba26174b (50)
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Content-Length: 0 (17)
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11:  (0)

[Mar 10 12:02:15] --- (11 headers 0 lines) ---
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:11239 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 52, ours 52)

[Mar 10 12:02:15] Using latest REGISTER request as basis request

[Mar 10 12:02:15] Sending to 192.168.1.97 : 5060 (NAT)

[Mar 10 12:02:15] Transmitting (NAT) to 192.168.1.97:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.97:5060;branch=z9hG4bK3278849582;received=192.168.1.97;rport=5060
From: "Display Name" <sip:551@24.123.23.170>;tag=4282629717
To: "Display Name" <sip:551@24.123.23.170>
Call-ID: 1319586744@192.168.1.97
CSeq: 52 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:551@24.123.23.170>
Content-Length: 0


---

[Mar 10 12:02:15] Transmitting (NAT) to 192.168.1.97:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.97:5060;branch=z9hG4bK3278849582;received=192.168.1.97;rport=5060
From: "Display Name" <sip:551@24.123.23.170>;tag=4282629717
To: "Display Name" <sip:551@24.123.23.170>;tag=as2e7b0a8c
Call-ID: 1319586744@192.168.1.97
CSeq: 52 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="30602214"
Content-Length: 0


---

[Mar 10 12:02:15] Scheduling destruction of call '1319586744@192.168.1.97' in 15000 ms

[Mar 10 12:02:15] 
<-- SIP read from 192.168.1.97:5060: 
REGISTER sip:24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.97:5060;rport;branch=z9hG4bK1372590648
From: "Display Name" <sip:551@24.123.23.170>;tag=366732303
To: "Display Name" <sip:551@24.123.23.170>
Call-ID: 1319586744@192.168.1.97
CSeq: 53 REGISTER
Contact: <sip:551@192.168.1.97:5060>;action=proxy
Authorization: Digest username="551", realm="asterisk", nonce="30602214", uri="sip:24.123.23.170", response="147c47a20af903fd9f41aad07d46cca2", algorithm=MD5
max-forwards: 70
expires: 60
user-agent: UTSTARCOM F1000/Device ID-0007ba26174b
Content-Length: 0


[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34)
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.97:5060;rport;branch=z9hG4bK1372590648 (65)
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: From: "Display Name" <sip:551@24.123.23.170>;tag=366732303 (58)
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: To: "Display Name" <sip:551@24.123.23.170> (42)
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Call-ID: 1319586744@192.168.1.97 (32)
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: CSeq: 53 REGISTER (17)
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: Contact: <sip:551@192.168.1.97:5060>;action=proxy (49)
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Authorization: Digest username="551", realm="asterisk", nonce="30602214", uri="sip:24.123.23.170", response="147c47a20af903fd9f41aad07d46cca2", algorithm=MD5 (157)
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: max-forwards: 70 (16)
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: expires: 60 (11)
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: user-agent: UTSTARCOM F1000/Device ID-0007ba26174b (50)
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11: Content-Length: 0 (17)
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 12:  (0)

[Mar 10 12:02:15] --- (12 headers 0 lines) ---
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:02:15] Using latest REGISTER request as basis request

[Mar 10 12:02:15] Sending to 192.168.1.97 : 5060 (NAT)

[Mar 10 12:02:15] Transmitting (NAT) to 192.168.1.97:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.97:5060;branch=z9hG4bK1372590648;received=192.168.1.97;rport=5060
From: "Display Name" <sip:551@24.123.23.170>;tag=366732303
To: "Display Name" <sip:551@24.123.23.170>
Call-ID: 1319586744@192.168.1.97
CSeq: 53 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:551@24.123.23.170>
Content-Length: 0


---

[Mar 10 12:02:15] Transmitting (NAT) to 192.168.1.97:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.97:5060;branch=z9hG4bK1372590648;received=192.168.1.97;rport=5060
From: "Display Name" <sip:551@24.123.23.170>;tag=366732303
To: "Display Name" <sip:551@24.123.23.170>;tag=as2e7b0a8c
Call-ID: 1319586744@192.168.1.97
CSeq: 53 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 60
Contact: <sip:551@192.168.1.97:5060>;expires=60
Date: Sat, 10 Mar 2007 17:02:15 GMT
Content-Length: 0


---

[Mar 10 12:02:15] Scheduling destruction of call '1319586744@192.168.1.97' in 15000 ms
[Mar 10 12:02:15] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 551
[Mar 10 12:02:15] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/551 - state 2 (In use)
[Mar 10 12:02:15] DEBUG[15208]: app_queue.c:500 changethread: Device 'SIP/551' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[Mar 10 12:02:15] DEBUG[15149]: chan_sip.c:1334 __sip_autodestruct: Auto destroying call '71f439ca643c96aa63da468e6d077ab4@24.123.23.170'

[Mar 10 12:02:15] Destroying call '71f439ca643c96aa63da468e6d077ab4@24.123.23.170'
[Mar 10 12:02:15] DEBUG[15195]: rtp.c:411 ast_rtcp_read: Got RTCP report of 64 bytes
[Mar 10 12:02:16] DEBUG[15209]: manager.c:1249 process_message: Manager received command 'Login'

[Mar 10 12:02:16]   == Parsing '/etc/asterisk/manager.conf': [Mar 10 12:02:16] DEBUG[15209]: config.c:595 config_text_file_load: Parsing /etc/asterisk/manager.conf
[Mar 10 12:02:16] Found
[Mar 10 12:02:16] DEBUG[15209]: acl.c:199 ast_append_ha: 127.0.0.1/255.255.255.0/255.255.255.0 appended to acl for peer
[Mar 10 12:02:16] DEBUG[15209]: acl.c:211 ast_apply_ha: ##### Testing 127.0.0.1 with 127.0.0.0

[Mar 10 12:02:16]   == Manager 'MessageNet' logged on from 127.0.0.1
[Mar 10 12:02:16] DEBUG[15209]: manager.c:1249 process_message: Manager received command 'Command'
[Mar 10 12:02:16] DEBUG[15209]: manager.c:1249 process_message: Manager received command 'Command'
[Mar 10 12:02:16] DEBUG[15209]: manager.c:1249 process_message: Manager received command 'Logoff'

[Mar 10 12:02:16]   == Manager 'MessageNet' logged off from 127.0.0.1

[Mar 10 12:02:17] 
<-- SIP read from 192.168.1.97:5060: 


[Mar 10 12:02:17] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0:  (0)

[Mar 10 12:02:17] --- (0 headers 0 lines) Nat keepalive ---

[Mar 10 12:02:18] 
<-- SIP read from 192.168.1.85:5060: 
INVITE sip:551@24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKxf1c8d2d21e430135f4b17fed3d2dbcfe
Call-ID: 3890f17d116db7343bb188d42c093fd2@24.123.23.170
CSeq: 509800 INVITE
From: 523 <sip:523@192.168.1.85:5060>;tag=a997c53df28f4a860026417db9e2f2ab
To: <sip:551@24.123.23.170>;tag=as6fedd0f9
Contact: <sip:523@192.168.1.85:5060>
Content-Type: application/sdp
User-Agent: Uniden SIP Phone p2 Ver BS4.77
Max-Forwards: 70
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK
Content-Length: 218

v=0
o=- 15131 298889 IN IP4 192.168.1.85
s=session
c=IN IP4 0.0.0.0
t=0 0
m=audio 30014 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,16,17,18
a=silenceSupp:off - - - -

[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: INVITE sip:551@24.123.23.170 SIP/2.0 (36)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKxf1c8d2d21e430135f4b17fed3d2dbcfe (82)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: Call-ID: 3890f17d116db7343bb188d42c093fd2@24.123.23.170 (55)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: CSeq: 509800 INVITE (19)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: From: 523 <sip:523@192.168.1.85:5060>;tag=a997c53df28f4a860026417db9e2f2ab (74)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: To: <sip:551@24.123.23.170>;tag=as6fedd0f9 (42)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: Contact: <sip:523@192.168.1.85:5060> (36)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Content-Type: application/sdp (29)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Max-Forwards: 70 (16)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 12: Content-Length: 218 (19)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 13:  (0)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: v=0 (3)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: o=- 15131 298889 IN IP4 192.168.1.85 (36)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: s=session (9)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: c=IN IP4 0.0.0.0 (16)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: t=0 0 (5)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: m=audio 30014 RTP/AVP 0 101 (27)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:0 PCMU/8000 (20)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=fmtp:101 0-15,16,17,18 (24)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=silenceSupp:off - - - - (25)

[Mar 10 12:02:18] --- (13 headers 10 lines) ---
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received INVITE (5) - Command in SIP INVITE
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:1015 parse_sip_options: Begin: parsing SIP "Supported: sip-cc, sip-cc-01, replaces, timer"
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:1027 parse_sip_options: Found SIP option: -sip-cc-
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:1038 parse_sip_options: Found no match for SIP option: sip-cc (Please file bug report!)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:1027 parse_sip_options: Found SIP option: -sip-cc-01-
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:1038 parse_sip_options: Found no match for SIP option: sip-cc-01 (Please file bug report!)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:1027 parse_sip_options: Found SIP option: -replaces-
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:1033 parse_sip_options: Matched SIP option: replaces
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:1027 parse_sip_options: Found SIP option: -timer-
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:1033 parse_sip_options: Matched SIP option: timer
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:1044 parse_sip_options: * SIP extension value: 5 for call 3890f17d116db7343bb188d42c093fd2@24.123.23.170

[Mar 10 12:02:18] Using INVITE request as basis request - 3890f17d116db7343bb188d42c093fd2@24.123.23.170

[Mar 10 12:02:18] Sending to 192.168.1.85 : 5060 (non-NAT)

[Mar 10 12:02:18] Found RTP audio format 0

[Mar 10 12:02:18] Found RTP audio format 101

[Mar 10 12:02:18] Peer audio RTP is at port 0.0.0.0:30014
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3631 process_sdp: Peer audio RTP is at port 0.0.0.0:30014

[Mar 10 12:02:18] Peer video RTP is at port 0.0.0.0:65535
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3653 process_sdp: Peer video RTP is at port 0.0.0.0:65535

[Mar 10 12:02:18] Found description format PCMU

[Mar 10 12:02:18] Found description format telephone-event

[Mar 10 12:02:18] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)

[Mar 10 12:02:18] Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Mar 10 12:02:18] DEBUG[15149]: channel.c:2409 set_format: Set channel SIP/551-08518cd8 to write format slin

[Mar 10 12:02:18]     -- Started music on hold, class 'default', on channel 'SIP/551-08518cd8'
[Mar 10 12:02:18] DEBUG[15149]: channel.c:1761 ast_settimeout: Scheduling timer at 160 sample intervals
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:10625 handle_request_invite: Got a SIP re-invite for call 3890f17d116db7343bb188d42c093fd2@24.123.23.170

[Mar 10 12:02:18] We're at 24.123.23.170 port 14572

[Mar 10 12:02:18] Video is at 24.123.23.170 port 16640

[Mar 10 12:02:18] Adding codec 0x4 (ulaw) to SDP

[Mar 10 12:02:18] Adding codec 0x8 (alaw) to SDP

[Mar 10 12:02:18] Adding codec 0x10 (g726) to SDP

[Mar 10 12:02:18] Adding codec 0x100 (g729) to SDP

[Mar 10 12:02:18] Adding non-codec 0x1 (telephone-event) to SDP

[Mar 10 12:02:18] Reliably Transmitting (no NAT) to 192.168.1.85:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKxf1c8d2d21e430135f4b17fed3d2dbcfe;received=192.168.1.85
From: 523 <sip:523@192.168.1.85:5060>;tag=a997c53df28f4a860026417db9e2f2ab
To: <sip:551@24.123.23.170>;tag=as6fedd0f9
Call-ID: 3890f17d116db7343bb188d42c093fd2@24.123.23.170
CSeq: 509800 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:551@24.123.23.170>
Content-Type: application/sdp
Content-Length: 318

v=0
o=root 15131 15133 IN IP4 192.168.1.97
s=session
c=IN IP4 192.168.1.97
t=0 0
m=audio 10010 RTP/AVP 0 8 111 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:1304 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id  #131
[Mar 10 12:02:18] DEBUG[15201]: rtp.c:1672 ast_rtp_bridge: Oooh, 'SIP/523-08521b58' changed end address to 0.0.0.0:30014 (format 4)
[Mar 10 12:02:18] DEBUG[15201]: rtp.c:1674 ast_rtp_bridge: Oooh, 'SIP/523-08521b58' changed end vaddress to 0.0.0.0:65535 (format 4)
[Mar 10 12:02:18] DEBUG[15201]: rtp.c:1676 ast_rtp_bridge: Oooh, 'SIP/523-08521b58' was 192.168.1.85:30014/(format 4)
[Mar 10 12:02:18] DEBUG[15201]: rtp.c:1678 ast_rtp_bridge: Oooh, 'SIP/523-08521b58' was 192.168.1.85:65535/(format 4)
[Mar 10 12:02:18] DEBUG[15201]: chan_sip.c:13028 sip_set_rtp_peer: Sending reinvite on SIP '3175191617@192.168.1.97' - It's audio soon redirected to IP 24.123.23.170

[Mar 10 12:02:18] set_destination: Parsing <sip:551@192.168.1.97:5060> for address/port to send to

[Mar 10 12:02:18] set_destination: set destination to 192.168.1.97, port 5060

[Mar 10 12:02:18] We're at 24.123.23.170 port 19120

[Mar 10 12:02:18] Video is at 24.123.23.170 port 13200

[Mar 10 12:02:18] Adding codec 0x4 (ulaw) to SDP

[Mar 10 12:02:18] Adding non-codec 0x1 (telephone-event) to SDP
[Mar 10 12:02:18] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 0: INVITE sip:551@192.168.1.97:5060 SIP/2.0 (40)
[Mar 10 12:02:18] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK4dd0d53e;rport (64)
[Mar 10 12:02:18] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 2: From: <sip:523@24.123.23.170>;tag=as57536955 (44)
[Mar 10 12:02:18] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 3: To: "Display Name" <sip:551@24.123.23.170>;tag=1341134544 (57)
[Mar 10 12:02:18] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 4: Contact: <sip:523@24.123.23.170> (32)
[Mar 10 12:02:18] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 5: Call-ID: 3175191617@192.168.1.97 (32)
[Mar 10 12:02:18] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 6: CSeq: 103 INVITE (16)
[Mar 10 12:02:18] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 7: User-Agent: Asterisk PBX (24)
[Mar 10 12:02:18] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 8: Max-Forwards: 70 (16)
[Mar 10 12:02:18] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66)
[Mar 10 12:02:18] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 10: X-asterisk-info: SIP re-invite (RTP bridge) (43)
[Mar 10 12:02:18] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 11: Content-Type: application/sdp (29)
[Mar 10 12:02:18] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 12: Content-Length: 218 (19)
[Mar 10 12:02:18] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 13:  (0)
[Mar 10 12:02:18] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: v=0 (3)
[Mar 10 12:02:18] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: o=root 15131 15133 IN IP4 24.123.23.170 (39)
[Mar 10 12:02:18] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: s=session (9)
[Mar 10 12:02:18] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: c=IN IP4 24.123.23.170 (22)
[Mar 10 12:02:18] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: t=0 0 (5)
[Mar 10 12:02:18] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: m=audio 19120 RTP/AVP 0 101 (27)
[Mar 10 12:02:18] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: a=rtpmap:0 PCMU/8000 (20)
[Mar 10 12:02:18] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33)
[Mar 10 12:02:18] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: a=fmtp:101 0-16 (15)
[Mar 10 12:02:18] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: a=silenceSupp:off - - - - (25)

[Mar 10 12:02:18] 13 headers, 10 lines

[Mar 10 12:02:18] Reliably Transmitting (NAT) to 192.168.1.97:5060:
INVITE sip:551@192.168.1.97:5060 SIP/2.0
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK4dd0d53e;rport
From: <sip:523@24.123.23.170>;tag=as57536955
To: "Display Name" <sip:551@24.123.23.170>;tag=1341134544
Contact: <sip:523@24.123.23.170>
Call-ID: 3175191617@192.168.1.97
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 218

v=0
o=root 15131 15133 IN IP4 24.123.23.170
s=session
c=IN IP4 24.123.23.170
t=0 0
m=audio 19120 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
[Mar 10 12:02:18] DEBUG[15201]: chan_sip.c:1304 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id  #132
[Mar 10 12:02:18] DEBUG[15201]: rtp.c:1359 ast_rtp_write: Ooh, format changed from unknown to ulaw

[Mar 10 12:02:18] 
<-- SIP read from 192.168.1.85:5060: 
ACK sip:551@24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKt6732da49670ec60bd472c930fa390d98
CSeq: 509800 ACK
To: <sip:551@24.123.23.170>;tag=as6fedd0f9
Call-ID: 3890f17d116db7343bb188d42c093fd2@24.123.23.170
From: 523 <sip:523@192.168.1.85:5060>;tag=a997c53df28f4a860026417db9e2f2ab
User-Agent: Uniden SIP Phone p2 Ver BS4.77


[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: ACK sip:551@24.123.23.170 SIP/2.0 (33)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKt6732da49670ec60bd472c930fa390d98 (82)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: CSeq: 509800 ACK (16)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: To: <sip:551@24.123.23.170>;tag=as6fedd0f9 (42)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Call-ID: 3890f17d116db7343bb188d42c093fd2@24.123.23.170 (55)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: From: 523 <sip:523@192.168.1.85:5060>;tag=a997c53df28f4a860026417db9e2f2ab (74)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7:  (0)

[Mar 10 12:02:18] --- (7 headers 0 lines) ---
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received ACK (6) - Command in SIP ACK
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:1401 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #131
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:1412 __sip_ack: Stopping retransmission on '3890f17d116db7343bb188d42c093fd2@24.123.23.170' of Response 509800: Match Found

[Mar 10 12:02:18] 
<-- SIP read from 192.168.1.97:5060: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK4dd0d53e;rport=5060
From: <sip:523@24.123.23.170>;tag=as57536955
To: "Display Name" <sip:551@24.123.23.170>;tag=1341134544
Call-ID: 3175191617@192.168.1.97
CSeq: 103 INVITE
Contact: <sip:551@192.168.1.97:5060>
Content-Type: application/sdp
Content-Length: 199  

v=0
o=551 123456 654331 IN IP4 192.168.1.97
s=none
c=IN IP4 192.168.1.97
t=0 0
m=audio 10010 RTP/AVP 0 101
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: SIP/2.0 200 OK (14)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK4dd0d53e;rport=5060 (69)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: From: <sip:523@24.123.23.170>;tag=as57536955 (44)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: To: "Display Name" <sip:551@24.123.23.170>;tag=1341134544 (57)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Call-ID: 3175191617@192.168.1.97 (32)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: CSeq: 103 INVITE (16)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: Contact: <sip:551@192.168.1.97:5060> (36)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Content-Type: application/sdp (29)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Content-Length: 199   (21)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9:  (0)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: v=0 (3)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: o=551 123456 654331 IN IP4 192.168.1.97 (39)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: s=none (6)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: c=IN IP4 192.168.1.97 (21)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: t=0 0 (5)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: m=audio 10010 RTP/AVP 0 101 (27)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=ptime:20 (10)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:0 PCMU/8000 (20)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=fmtp:101 0-16 (15)

[Mar 10 12:02:18] --- (9 headers 10 lines) ---
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:1390 __sip_ack: Acked pending invite 103
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:1401 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #132
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:1412 __sip_ack: Stopping retransmission on '3175191617@192.168.1.97' of Request 103: Match Found
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:9640 handle_response_invite: SIP response 200 to RE-invite on outgoing call 3175191617@192.168.1.97

[Mar 10 12:02:18] Found RTP audio format 0

[Mar 10 12:02:18] Found RTP audio format 101

[Mar 10 12:02:18] Peer audio RTP is at port 192.168.1.97:10010
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3631 process_sdp: Peer audio RTP is at port 192.168.1.97:10010

[Mar 10 12:02:18] Peer video RTP is at port 192.168.1.97:65535
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3653 process_sdp: Peer video RTP is at port 192.168.1.97:65535

[Mar 10 12:02:18] Found description format PCMU

[Mar 10 12:02:18] Found description format telephone-event

[Mar 10 12:02:18] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)

[Mar 10 12:02:18] Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:6137 build_route: build_route: Retaining previous route: <sip:551@192.168.1.97:5060>

[Mar 10 12:02:18] set_destination: Parsing <sip:551@192.168.1.97:5060> for address/port to send to

[Mar 10 12:02:18] set_destination: set destination to 192.168.1.97, port 5060

[Mar 10 12:02:18] Transmitting (NAT) to 192.168.1.97:5060:
ACK sip:551@192.168.1.97:5060 SIP/2.0
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK68d4fd2a;rport
From: <sip:523@24.123.23.170>;tag=as57536955
To: "Display Name" <sip:551@24.123.23.170>;tag=1341134544
Contact: <sip:523@24.123.23.170>
Call-ID: 3175191617@192.168.1.97
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---

[Mar 10 12:02:18] 
<-- SIP read from 192.168.1.85:5060: 
INVITE sip:593@24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKaf8142956f618711e28805bf4c5289dfc
Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170
CSeq: 388769 INVITE
From: 523 <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f
To: <sip:593@24.123.23.170>;tag=as0da2dd59
Contact: <sip:523@192.168.1.85:5060>
Content-Type: application/sdp
User-Agent: Uniden SIP Phone p2 Ver BS4.77
Max-Forwards: 70
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK
Content-Length: 269

v=0
o=- 1790166131 248058 IN IP4 192.168.1.85
s=-
c=IN IP4 192.168.1.85
t=0 0
m=audio 30012 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,16,17,18
a=sendrecv
a=ptime:20

[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: INVITE sip:593@24.123.23.170 SIP/2.0 (36)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKaf8142956f618711e28805bf4c5289dfc (82)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170 (55)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: CSeq: 388769 INVITE (19)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: From: 523 <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f (74)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: To: <sip:593@24.123.23.170>;tag=as0da2dd59 (42)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: Contact: <sip:523@192.168.1.85:5060> (36)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Content-Type: application/sdp (29)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Max-Forwards: 70 (16)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 12: Content-Length: 269 (19)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 13:  (0)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: v=0 (3)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: o=- 1790166131 248058 IN IP4 192.168.1.85 (41)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: s=- (3)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: c=IN IP4 192.168.1.85 (21)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: t=0 0 (5)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: m=audio 30012 RTP/AVP 0 8 18 101 (32)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:0 PCMU/8000 (20)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:8 PCMA/8000 (20)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:18 G729/8000 (21)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=fmtp:101 0-15,16,17,18 (24)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=sendrecv (10)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=ptime:20 (10)

[Mar 10 12:02:18] --- (13 headers 13 lines) ---
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received INVITE (5) - Command in SIP INVITE

[Mar 10 12:02:18] Using INVITE request as basis request - 5d28f46f54fd450475a9f02419f99674@24.123.23.170

[Mar 10 12:02:18] Sending to 192.168.1.85 : 5060 (non-NAT)

[Mar 10 12:02:18] Found RTP audio format 0

[Mar 10 12:02:18] Found RTP audio format 8

[Mar 10 12:02:18] Found RTP audio format 18

[Mar 10 12:02:18] Found RTP audio format 101

[Mar 10 12:02:18] Peer audio RTP is at port 192.168.1.85:30012
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3631 process_sdp: Peer audio RTP is at port 192.168.1.85:30012

[Mar 10 12:02:18] Peer video RTP is at port 192.168.1.85:65535
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3653 process_sdp: Peer video RTP is at port 192.168.1.85:65535

[Mar 10 12:02:18] Found description format PCMU

[Mar 10 12:02:18] Found description format PCMA

[Mar 10 12:02:18] Found description format G729

[Mar 10 12:02:18] Found description format telephone-event

[Mar 10 12:02:18] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)

[Mar 10 12:02:18] Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Mar 10 12:02:18] DEBUG[15149]: channel.c:2409 set_format: Set channel SIP/593-084f44b8 to write format ulaw

[Mar 10 12:02:18]     -- Stopped music on hold on SIP/593-084f44b8
[Mar 10 12:02:18] DEBUG[15149]: channel.c:1761 ast_settimeout: Scheduling timer at 0 sample intervals
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:10625 handle_request_invite: Got a SIP re-invite for call 5d28f46f54fd450475a9f02419f99674@24.123.23.170

[Mar 10 12:02:18] We're at 24.123.23.170 port 17986

[Mar 10 12:02:18] Video is at 24.123.23.170 port 12600

[Mar 10 12:02:18] Adding codec 0x4 (ulaw) to SDP

[Mar 10 12:02:18] Adding codec 0x8 (alaw) to SDP

[Mar 10 12:02:18] Adding codec 0x100 (g729) to SDP

[Mar 10 12:02:18] Adding non-codec 0x1 (telephone-event) to SDP

[Mar 10 12:02:18] Reliably Transmitting (no NAT) to 192.168.1.85:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKaf8142956f618711e28805bf4c5289dfc;received=192.168.1.85
From: 523 <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f
To: <sip:593@24.123.23.170>;tag=as0da2dd59
Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170
CSeq: 388769 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:593@24.123.23.170>
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 15131 15134 IN IP4 192.168.1.83
s=session
c=IN IP4 192.168.1.83
t=0 0
m=audio 51000 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:1304 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id  #133
[Mar 10 12:02:18] DEBUG[15195]: rtp.c:1672 ast_rtp_bridge: Oooh, 'SIP/523-0850b038' changed end address to 192.168.1.85:30012 (format 268)
[Mar 10 12:02:18] DEBUG[15195]: rtp.c:1674 ast_rtp_bridge: Oooh, 'SIP/523-0850b038' changed end vaddress to 192.168.1.85:65535 (format 268)
[Mar 10 12:02:18] DEBUG[15195]: rtp.c:1676 ast_rtp_bridge: Oooh, 'SIP/523-0850b038' was 0.0.0.0:30012/(format 4)
[Mar 10 12:02:18] DEBUG[15195]: rtp.c:1678 ast_rtp_bridge: Oooh, 'SIP/523-0850b038' was 0.0.0.0:65535/(format 4)
[Mar 10 12:02:18] DEBUG[15195]: chan_sip.c:13028 sip_set_rtp_peer: Sending reinvite on SIP '20a99672af308312d36264481f6b7f5a@192.168.1.83' - It's audio soon redirected to IP 192.168.1.85

[Mar 10 12:02:18] set_destination: Parsing <sip:593@192.168.1.83:5060> for address/port to send to

[Mar 10 12:02:18] set_destination: set destination to 192.168.1.83, port 5060

[Mar 10 12:02:18] We're at 24.123.23.170 port 19424

[Mar 10 12:02:18] Video is at 24.123.23.170 port 13400

[Mar 10 12:02:18] Adding codec 0x4 (ulaw) to SDP

[Mar 10 12:02:18] Adding codec 0x8 (alaw) to SDP

[Mar 10 12:02:18] Adding codec 0x100 (g729) to SDP

[Mar 10 12:02:18] Adding non-codec 0x1 (telephone-event) to SDP
[Mar 10 12:02:18] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 0: INVITE sip:593@192.168.1.83:5060 SIP/2.0 (40)
[Mar 10 12:02:18] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK3bfc7bfe (58)
[Mar 10 12:02:18] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 2: From: 523 <sip:523@24.123.23.170:5060>;tag=as38239579 (53)
[Mar 10 12:02:18] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 3: To: 593 <sip:593@24.123.23.170:5060>;tag=b3eeb4c9225baf5 (56)
[Mar 10 12:02:18] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 4: Contact: <sip:523@24.123.23.170> (32)
[Mar 10 12:02:18] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 5: Call-ID: 20a99672af308312d36264481f6b7f5a@192.168.1.83 (54)
[Mar 10 12:02:18] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 6: CSeq: 104 INVITE (16)
[Mar 10 12:02:18] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 7: User-Agent: Asterisk PBX (24)
[Mar 10 12:02:18] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 8: Max-Forwards: 70 (16)
[Mar 10 12:02:18] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66)
[Mar 10 12:02:18] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 10: X-asterisk-info: SIP re-invite (RTP bridge) (43)
[Mar 10 12:02:18] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 11: Content-Type: application/sdp (29)
[Mar 10 12:02:18] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 12: Content-Length: 287 (19)
[Mar 10 12:02:18] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 13:  (0)
[Mar 10 12:02:18] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: v=0 (3)
[Mar 10 12:02:18] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: o=root 15131 15134 IN IP4 192.168.1.85 (38)
[Mar 10 12:02:18] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: s=session (9)
[Mar 10 12:02:18] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: c=IN IP4 192.168.1.85 (21)
[Mar 10 12:02:18] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: t=0 0 (5)
[Mar 10 12:02:18] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: m=audio 30012 RTP/AVP 0 8 18 101 (32)
[Mar 10 12:02:18] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: a=rtpmap:0 PCMU/8000 (20)
[Mar 10 12:02:18] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: a=rtpmap:8 PCMA/8000 (20)
[Mar 10 12:02:18] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: a=rtpmap:18 G729/8000 (21)
[Mar 10 12:02:18] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: a=fmtp:18 annexb=no (19)
[Mar 10 12:02:18] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33)
[Mar 10 12:02:18] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: a=fmtp:101 0-16 (15)
[Mar 10 12:02:18] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: a=silenceSupp:off - - - - (25)

[Mar 10 12:02:18] 13 headers, 13 lines

[Mar 10 12:02:18] Reliably Transmitting (no NAT) to 192.168.1.83:5060:
INVITE sip:593@192.168.1.83:5060 SIP/2.0
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK3bfc7bfe
From: 523 <sip:523@24.123.23.170:5060>;tag=as38239579
To: 593 <sip:593@24.123.23.170:5060>;tag=b3eeb4c9225baf5
Contact: <sip:523@24.123.23.170>
Call-ID: 20a99672af308312d36264481f6b7f5a@192.168.1.83
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 15131 15134 IN IP4 192.168.1.85
s=session
c=IN IP4 192.168.1.85
t=0 0
m=audio 30012 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
[Mar 10 12:02:18] DEBUG[15195]: chan_sip.c:1304 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id  #134
[Mar 10 12:02:18] DEBUG[15195]: rtp.c:1359 ast_rtp_write: Ooh, format changed from unknown to ulaw

[Mar 10 12:02:18] 
<-- SIP read from 192.168.1.93:5060: 
REGISTER sip:24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKlf10b70597a594d3edfc66fac651a08c4
Call-ID: 55ae66469f7f@24.123.23.170
CSeq: 48238 REGISTER
From: <sip:528@24.123.23.170>;tag=21504f1f403fa90232dd28e22a3fc5d4
To: <sip:528@24.123.23.170>
Contact: <sip:528@192.168.1.93:5060>;expires=60;q=0.50
User-Agent: Uniden SIP Phone p2 Ver BS4.77
Max-Forwards: 70
Content-Length: 0
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK
Authorization: Digest realm="asterisk",   nonce="244fa650",   algorithm=MD5,   uri="sip:24.123.23.170",   username="528",   response="30665ff0029279495862b32150a4ee76"


[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKlf10b70597a594d3edfc66fac651a08c4 (82)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: Call-ID: 55ae66469f7f@24.123.23.170 (35)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: CSeq: 48238 REGISTER (20)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: From: <sip:528@24.123.23.170>;tag=21504f1f403fa90232dd28e22a3fc5d4 (66)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: To: <sip:528@24.123.23.170> (27)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: Contact: <sip:528@192.168.1.93:5060>;expires=60;q=0.50 (54)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Max-Forwards: 70 (16)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Content-Length: 0 (17)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 12: Authorization: Digest realm="asterisk",   nonce="244fa650",   algorithm=MD5,   uri="sip:24.123.23.170",   username="528",   response="30665ff0029279495862b32150a4ee76" (167)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 13:  (0)

[Mar 10 12:02:18] --- (13 headers 0 lines) ---
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3168 sip_alloc: Allocating new SIP dialog for 55ae66469f7f@24.123.23.170 - REGISTER (No RTP)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:02:18] Using latest REGISTER request as basis request

[Mar 10 12:02:18] Sending to 192.168.1.93 : 5060 (non-NAT)

[Mar 10 12:02:18] Transmitting (no NAT) to 192.168.1.93:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKlf10b70597a594d3edfc66fac651a08c4;received=192.168.1.93
From: <sip:528@24.123.23.170>;tag=21504f1f403fa90232dd28e22a3fc5d4
To: <sip:528@24.123.23.170>
Call-ID: 55ae66469f7f@24.123.23.170
CSeq: 48238 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:528@24.123.23.170>
Content-Length: 0


---

[Mar 10 12:02:18] Transmitting (no NAT) to 192.168.1.93:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKlf10b70597a594d3edfc66fac651a08c4;received=192.168.1.93
From: <sip:528@24.123.23.170>;tag=21504f1f403fa90232dd28e22a3fc5d4
To: <sip:528@24.123.23.170>;tag=as4cd451ac
Call-ID: 55ae66469f7f@24.123.23.170
CSeq: 48238 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="66cde820"
Content-Length: 0


---

[Mar 10 12:02:18] Scheduling destruction of call '55ae66469f7f@24.123.23.170' in 15000 ms

[Mar 10 12:02:18] 
<-- SIP read from 192.168.1.83:5060: 
SIP/2.0 200 OK
Call-ID: 20a99672af308312d36264481f6b7f5a@192.168.1.83
CSeq: 104 INVITE
From: 523 <sip:523@24.123.23.170:5060>;tag=as38239579
To: 593 <sip:593@24.123.23.170:5060>;tag=b3eeb4c9225baf5
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK3bfc7bfe
Content-Length: 280
Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE
Content-Type: application/sdp
Supported: replaces
Contact: 593 <sip:593@192.168.1.83:5060>
User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4

v=0
o=MxSIP 0 1158539950 IN IP4 192.168.1.83
s=SIP Call
c=IN IP4 192.168.1.83
t=0 0
m=audio 51000 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:on - - - -

[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: SIP/2.0 200 OK (14)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Call-ID: 20a99672af308312d36264481f6b7f5a@192.168.1.83 (54)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: CSeq: 104 INVITE (16)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: From: 523 <sip:523@24.123.23.170:5060>;tag=as38239579 (53)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: To: 593 <sip:593@24.123.23.170:5060>;tag=b3eeb4c9225baf5 (56)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK3bfc7bfe (58)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: Content-Length: 280 (19)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE (55)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Content-Type: application/sdp (29)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Supported: replaces (19)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Contact: 593 <sip:593@192.168.1.83:5060> (40)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11: User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 (42)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 12:  (0)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: v=0 (3)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: o=MxSIP 0 1158539950 IN IP4 192.168.1.83 (40)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: s=SIP Call (10)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: c=IN IP4 192.168.1.83 (21)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: t=0 0 (5)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: m=audio 51000 RTP/AVP 0 8 18 101 (32)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:0 PCMU/8000 (20)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:8 PCMA/8000 (20)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:18 G729/8000 (21)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=fmtp:101 0-15 (15)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=ptime:20 (10)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=silenceSupp:on - - - - (24)

[Mar 10 12:02:18] --- (12 headers 13 lines) ---
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:1390 __sip_ack: Acked pending invite 104
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:1401 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #134
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:1412 __sip_ack: Stopping retransmission on '20a99672af308312d36264481f6b7f5a@192.168.1.83' of Request 104: Match Found
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:9640 handle_response_invite: SIP response 200 to RE-invite on outgoing call 20a99672af308312d36264481f6b7f5a@192.168.1.83

[Mar 10 12:02:18] Found RTP audio format 0

[Mar 10 12:02:18] Found RTP audio format 8

[Mar 10 12:02:18] Found RTP audio format 18

[Mar 10 12:02:18] Found RTP audio format 101

[Mar 10 12:02:18] Peer audio RTP is at port 192.168.1.83:51000
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3631 process_sdp: Peer audio RTP is at port 192.168.1.83:51000

[Mar 10 12:02:18] Peer video RTP is at port 192.168.1.83:65535
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3653 process_sdp: Peer video RTP is at port 192.168.1.83:65535

[Mar 10 12:02:18] Found description format PCMU

[Mar 10 12:02:18] Found description format PCMA

[Mar 10 12:02:18] Found description format G729

[Mar 10 12:02:18] Found description format telephone-event

[Mar 10 12:02:18] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)

[Mar 10 12:02:18] Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:6137 build_route: build_route: Retaining previous route: <sip:593@192.168.1.83:5060>

[Mar 10 12:02:18] set_destination: Parsing <sip:593@192.168.1.83:5060> for address/port to send to

[Mar 10 12:02:18] set_destination: set destination to 192.168.1.83, port 5060

[Mar 10 12:02:18] Transmitting (no NAT) to 192.168.1.83:5060:
ACK sip:593@192.168.1.83:5060 SIP/2.0
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK3a45b419
From: 523 <sip:523@24.123.23.170:5060>;tag=as38239579
To: 593 <sip:593@24.123.23.170:5060>;tag=b3eeb4c9225baf5
Contact: <sip:523@24.123.23.170>
Call-ID: 20a99672af308312d36264481f6b7f5a@192.168.1.83
CSeq: 104 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---

[Mar 10 12:02:18] 
<-- SIP read from 192.168.1.98:5060: 
REGISTER sip:24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.98:5060;branch=z9hG4bKb2954eef30da80546f4e6f4731d501c1b
Call-ID: 55ae664696c0@24.123.23.170
CSeq: 43029 REGISTER
From: <sip:529@24.123.23.170>;tag=52121cc114553c9be6eeeece2042eccc
To: <sip:529@24.123.23.170>
Contact: <sip:529@192.168.1.98:5060>;expires=60;q=0.50
User-Agent: Uniden SIP Phone p2 Ver BS4.77
Max-Forwards: 70
Content-Length: 0
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK
Authorization: Digest realm="asterisk",   nonce="57f968ff",   algorithm=MD5,   uri="sip:24.123.23.170",   username="529",   response="7e7f0d9e46d9fa4e5c7aea78983535cd"


[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.98:5060;branch=z9hG4bKb2954eef30da80546f4e6f4731d501c1b (82)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: Call-ID: 55ae664696c0@24.123.23.170 (35)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: CSeq: 43029 REGISTER (20)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: From: <sip:529@24.123.23.170>;tag=52121cc114553c9be6eeeece2042eccc (66)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: To: <sip:529@24.123.23.170> (27)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: Contact: <sip:529@192.168.1.98:5060>;expires=60;q=0.50 (54)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Max-Forwards: 70 (16)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Content-Length: 0 (17)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 12: Authorization: Digest realm="asterisk",   nonce="57f968ff",   algorithm=MD5,   uri="sip:24.123.23.170",   username="529",   response="7e7f0d9e46d9fa4e5c7aea78983535cd" (167)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 13:  (0)

[Mar 10 12:02:18] --- (13 headers 0 lines) ---
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3168 sip_alloc: Allocating new SIP dialog for 55ae664696c0@24.123.23.170 - REGISTER (No RTP)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:02:18] Using latest REGISTER request as basis request

[Mar 10 12:02:18] Sending to 192.168.1.98 : 5060 (non-NAT)

[Mar 10 12:02:18] Transmitting (no NAT) to 192.168.1.98:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.98:5060;branch=z9hG4bKb2954eef30da80546f4e6f4731d501c1b;received=192.168.1.98
From: <sip:529@24.123.23.170>;tag=52121cc114553c9be6eeeece2042eccc
To: <sip:529@24.123.23.170>
Call-ID: 55ae664696c0@24.123.23.170
CSeq: 43029 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:529@24.123.23.170>
Content-Length: 0


---

[Mar 10 12:02:18] Transmitting (no NAT) to 192.168.1.98:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.98:5060;branch=z9hG4bKb2954eef30da80546f4e6f4731d501c1b;received=192.168.1.98
From: <sip:529@24.123.23.170>;tag=52121cc114553c9be6eeeece2042eccc
To: <sip:529@24.123.23.170>;tag=as3601d8f0
Call-ID: 55ae664696c0@24.123.23.170
CSeq: 43029 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="061034be"
Content-Length: 0


---

[Mar 10 12:02:18] Scheduling destruction of call '55ae664696c0@24.123.23.170' in 15000 ms

[Mar 10 12:02:18] 
<-- SIP read from 192.168.1.93:5060: 
REGISTER sip:24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKi64fb7fb2ccc34264aa5dd147f60ffe13
CSeq: 48239 REGISTER
Call-ID: 55ae66469f7f@24.123.23.170
From: <sip:528@24.123.23.170>;tag=21504f1f403fa90232dd28e22a3fc5d4
To: <sip:528@24.123.23.170>
Contact: <sip:528@192.168.1.93:5060>;expires=60;q=0.50
User-Agent: Uniden SIP Phone p2 Ver BS4.77
Max-Forwards: 70
Content-Length: 0
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK
Authorization: Digest realm="asterisk",   nonce="66cde820",   algorithm=MD5,   uri="sip:24.123.23.170",   username="528",   response="4815c0d39f9b9684e4e1ca58892c62c1"


[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKi64fb7fb2ccc34264aa5dd147f60ffe13 (82)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: CSeq: 48239 REGISTER (20)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: Call-ID: 55ae66469f7f@24.123.23.170 (35)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: From: <sip:528@24.123.23.170>;tag=21504f1f403fa90232dd28e22a3fc5d4 (66)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: To: <sip:528@24.123.23.170> (27)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: Contact: <sip:528@192.168.1.93:5060>;expires=60;q=0.50 (54)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Max-Forwards: 70 (16)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Content-Length: 0 (17)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 12: Authorization: Digest realm="asterisk",   nonce="66cde820",   algorithm=MD5,   uri="sip:24.123.23.170",   username="528",   response="4815c0d39f9b9684e4e1ca58892c62c1" (167)
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 13:  (0)

[Mar 10 12:02:18] --- (13 headers 0 lines) ---
[Mar 10 12:02:18] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:02:18] Using latest REGISTER request as basis request

[Mar 10 12:02:18] Sending to 192.168.1.93 : 5060 (non-NAT)

[Mar 10 12:02:18] Transmitting (no NAT) to 192.168.1.93:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKi64fb7fb2ccc34264aa5dd147f60ffe13;received=192.168.1.93
From: <sip:528@24.123.23.170>;tag=21504f1f403fa90232dd28e22a3fc5d4
To: <sip:528@24.123.23.170>
Call-ID: 55ae66469f7f@24.123.23.170
CSeq: 48239 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:528@24.123.23.170>
Content-Length: 0


---

[Mar 10 12:02:18] Transmitting (no NAT) to 192.168.1.93:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKi64fb7fb2ccc34264aa5dd147f60ffe13;received=192.168.1.93
From: <sip:528@24.123.23.170>;tag=21504f1f403fa90232dd28e22a3fc5d4
To: <sip:528@24.123.23.170>;tag=as4cd451ac
Call-ID: 55ae66469f7f@24.123.23.170
CSeq: 48239 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 60
Contact: <sip:528@192.168.1.93:5060>;expires=60
Date: Sat, 10 Mar 2007 17:02:18 GMT
Content-Length: 0


---

[Mar 10 12:02:18] Scheduling destruction of call '55ae66469f7f@24.123.23.170' in 15000 ms
[Mar 10 12:02:18] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 528
[Mar 10 12:02:18] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/528 - state 1 (Not in use)
[Mar 10 12:02:18] DEBUG[15210]: app_queue.c:500 changethread: Device 'SIP/528' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.

[Mar 10 12:02:19] 
<-- SIP read from 192.168.1.85:5060: 
INVITE sip:593@24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKaf8142956f618711e28805bf4c5289dfc
Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170
CSeq: 388769 INVITE
From: 523 <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f
To: <sip:593@24.123.23.170>;tag=as0da2dd59
Contact: <sip:523@192.168.1.85:5060>
Content-Type: application/sdp
User-Agent: Uniden SIP Phone p2 Ver BS4.77
Max-Forwards: 70
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK
Content-Length: 269

v=0
o=- 1790166131 248058 IN IP4 192.168.1.85
s=-
c=IN IP4 192.168.1.85
t=0 0
m=audio 30012 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,16,17,18
a=sendrecv
a=ptime:20

[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: INVITE sip:593@24.123.23.170 SIP/2.0 (36)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKaf8142956f618711e28805bf4c5289dfc (82)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170 (55)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: CSeq: 388769 INVITE (19)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: From: 523 <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f (74)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: To: <sip:593@24.123.23.170>;tag=as0da2dd59 (42)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: Contact: <sip:523@192.168.1.85:5060> (36)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Content-Type: application/sdp (29)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Max-Forwards: 70 (16)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 12: Content-Length: 269 (19)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 13:  (0)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: v=0 (3)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: o=- 1790166131 248058 IN IP4 192.168.1.85 (41)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: s=- (3)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: c=IN IP4 192.168.1.85 (21)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: t=0 0 (5)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: m=audio 30012 RTP/AVP 0 8 18 101 (32)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:0 PCMU/8000 (20)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:8 PCMA/8000 (20)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:18 G729/8000 (21)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=fmtp:101 0-15,16,17,18 (24)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=sendrecv (10)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=ptime:20 (10)

[Mar 10 12:02:19] --- (13 headers 13 lines) ---
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received INVITE (5) - Command in SIP INVITE
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:11239 handle_request: Ignoring SIP message because of retransmit (INVITE Seqno 388769, ours 388769)

[Mar 10 12:02:19] Ignoring this INVITE request
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:10625 handle_request_invite: Got a SIP re-invite for call 5d28f46f54fd450475a9f02419f99674@24.123.23.170

[Mar 10 12:02:19] We're at 24.123.23.170 port 17986

[Mar 10 12:02:19] Video is at 24.123.23.170 port 12600

[Mar 10 12:02:19] Adding codec 0x4 (ulaw) to SDP

[Mar 10 12:02:19] Adding codec 0x8 (alaw) to SDP

[Mar 10 12:02:19] Adding codec 0x100 (g729) to SDP

[Mar 10 12:02:19] Adding non-codec 0x1 (telephone-event) to SDP

[Mar 10 12:02:19] Reliably Transmitting (no NAT) to 192.168.1.85:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKaf8142956f618711e28805bf4c5289dfc;received=192.168.1.85
From: 523 <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f
To: <sip:593@24.123.23.170>;tag=as0da2dd59
Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170
CSeq: 388769 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:593@24.123.23.170>
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 15131 15135 IN IP4 192.168.1.83
s=session
c=IN IP4 192.168.1.83
t=0 0
m=audio 51000 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:1304 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id  #139

[Mar 10 12:02:19] 
<-- SIP read from 192.168.1.98:5060: 
REGISTER sip:24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.98:5060;branch=z9hG4bKg204c54a76bfdc2a0d470dbbe6c08d81f
CSeq: 43030 REGISTER
Call-ID: 55ae664696c0@24.123.23.170
From: <sip:529@24.123.23.170>;tag=52121cc114553c9be6eeeece2042eccc
To: <sip:529@24.123.23.170>
Contact: <sip:529@192.168.1.98:5060>;expires=60;q=0.50
User-Agent: Uniden SIP Phone p2 Ver BS4.77
Max-Forwards: 70
Content-Length: 0
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK
Authorization: Digest realm="asterisk",   nonce="061034be",   algorithm=MD5,   uri="sip:24.123.23.170",   username="529",   response="862546c3420c3b4d2bc99ecfcc1e69d8"


[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.98:5060;branch=z9hG4bKg204c54a76bfdc2a0d470dbbe6c08d81f (82)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: CSeq: 43030 REGISTER (20)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: Call-ID: 55ae664696c0@24.123.23.170 (35)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: From: <sip:529@24.123.23.170>;tag=52121cc114553c9be6eeeece2042eccc (66)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: To: <sip:529@24.123.23.170> (27)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: Contact: <sip:529@192.168.1.98:5060>;expires=60;q=0.50 (54)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Max-Forwards: 70 (16)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Content-Length: 0 (17)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 12: Authorization: Digest realm="asterisk",   nonce="061034be",   algorithm=MD5,   uri="sip:24.123.23.170",   username="529",   response="862546c3420c3b4d2bc99ecfcc1e69d8" (167)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 13:  (0)

[Mar 10 12:02:19] --- (13 headers 0 lines) ---
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:02:19] Using latest REGISTER request as basis request

[Mar 10 12:02:19] Sending to 192.168.1.98 : 5060 (non-NAT)

[Mar 10 12:02:19] Transmitting (no NAT) to 192.168.1.98:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.98:5060;branch=z9hG4bKg204c54a76bfdc2a0d470dbbe6c08d81f;received=192.168.1.98
From: <sip:529@24.123.23.170>;tag=52121cc114553c9be6eeeece2042eccc
To: <sip:529@24.123.23.170>
Call-ID: 55ae664696c0@24.123.23.170
CSeq: 43030 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:529@24.123.23.170>
Content-Length: 0


---

[Mar 10 12:02:19] Transmitting (no NAT) to 192.168.1.98:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.98:5060;branch=z9hG4bKg204c54a76bfdc2a0d470dbbe6c08d81f;received=192.168.1.98
From: <sip:529@24.123.23.170>;tag=52121cc114553c9be6eeeece2042eccc
To: <sip:529@24.123.23.170>;tag=as3601d8f0
Call-ID: 55ae664696c0@24.123.23.170
CSeq: 43030 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 60
Contact: <sip:529@192.168.1.98:5060>;expires=60
Date: Sat, 10 Mar 2007 17:02:19 GMT
Content-Length: 0


---

[Mar 10 12:02:19] Scheduling destruction of call '55ae664696c0@24.123.23.170' in 15000 ms
[Mar 10 12:02:19] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 529
[Mar 10 12:02:19] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/529 - state 1 (Not in use)
[Mar 10 12:02:19] DEBUG[15211]: app_queue.c:500 changethread: Device 'SIP/529' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:02:19] DEBUG[15201]: rtp.c:411 ast_rtcp_read: Got RTCP report of 64 bytes

[Mar 10 12:02:19] 
<-- SIP read from 192.168.1.85:5060: 
ACK sip:593@24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKg5a795cae7097e5e99fa3c05829cb73d8
CSeq: 388769 ACK
To: <sip:593@24.123.23.170>;tag=as0da2dd59
Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170
From: 523 <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f
User-Agent: Uniden SIP Phone p2 Ver BS4.77


[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: ACK sip:593@24.123.23.170 SIP/2.0 (33)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKg5a795cae7097e5e99fa3c05829cb73d8 (82)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: CSeq: 388769 ACK (16)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: To: <sip:593@24.123.23.170>;tag=as0da2dd59 (42)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170 (55)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: From: 523 <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f (74)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42)
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7:  (0)

[Mar 10 12:02:19] --- (7 headers 0 lines) ---
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received ACK (6) - Command in SIP ACK
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:1401 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #139
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:1412 __sip_ack: Stopping retransmission on '5d28f46f54fd450475a9f02419f99674@24.123.23.170' of Response 388769: Match Found
[Mar 10 12:02:19] DEBUG[15201]: rtp.c:502 ast_rtp_read: Forcing Marker bit, because SSRC has changed
[Mar 10 12:02:19] DEBUG[15201]: channel.c:2027 ast_read: Generator got voice, switching to phase locked mode
[Mar 10 12:02:19] DEBUG[15201]: channel.c:1761 ast_settimeout: Scheduling timer at 0 sample intervals
[Mar 10 12:02:19] DEBUG[15201]: rtp.c:411 ast_rtcp_read: Got RTCP report of 40 bytes
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:1195 retrans_pkt: SIP TIMER: Rescheduling retransmission #133 (1) SIP/2.0 - 1
[Mar 10 12:02:19] DEBUG[15149]: chan_sip.c:1209 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #133)) 

[Mar 10 12:02:19] Retransmitting #1 (no NAT) to 192.168.1.85:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKaf8142956f618711e28805bf4c5289dfc;received=192.168.1.85
From: 523 <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f
To: <sip:593@24.123.23.170>;tag=as0da2dd59
Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170
CSeq: 388769 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:593@24.123.23.170>
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 15131 15134 IN IP4 192.168.1.83
s=session
c=IN IP4 192.168.1.83
t=0 0
m=audio 51000 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
[Mar 10 12:02:19] DEBUG[15137]: res_musiconhold.c:549 monmp3thread: Only wrote -1 of 640 bytes to pipe
[Mar 10 12:02:19] DEBUG[15137]: res_musiconhold.c:549 monmp3thread: Only wrote -1 of 640 bytes to pipe
[Mar 10 12:02:19] DEBUG[15137]: res_musiconhold.c:549 monmp3thread: Only wrote -1 of 640 bytes to pipe
[Mar 10 12:02:19] DEBUG[15137]: res_musiconhold.c:549 monmp3thread: Only wrote -1 of 640 bytes to pipe
[Mar 10 12:02:19] DEBUG[15137]: res_musiconhold.c:549 monmp3thread: Only wrote -1 of 640 bytes to pipe
[Mar 10 12:02:19] DEBUG[15137]: res_musiconhold.c:549 monmp3thread: Only wrote -1 of 640 bytes to pipe
[Mar 10 12:02:19] DEBUG[15137]: res_musiconhold.c:549 monmp3thread: Only wrote -1 of 640 bytes to pipe
[Mar 10 12:02:19] DEBUG[15137]: res_musiconhold.c:549 monmp3thread: Only wrote -1 of 640 bytes to pipe
[Mar 10 12:02:19] DEBUG[15137]: res_musiconhold.c:549 monmp3thread: Only wrote -1 of 640 bytes to pipe
[Mar 10 12:02:19] DEBUG[15137]: res_musiconhold.c:549 monmp3thread: Only wrote -1 of 640 bytes to pipe
[Mar 10 12:02:19] DEBUG[15137]: res_musiconhold.c:549 monmp3thread: Only wrote -1 of 640 bytes to pipe
[Mar 10 12:02:19] DEBUG[15137]: res_musiconhold.c:549 monmp3thread: Only wrote -1 of 640 bytes to pipe
[Mar 10 12:02:20] DEBUG[15137]: res_musiconhold.c:549 monmp3thread: Only wrote -1 of 640 bytes to pipe

[Mar 10 12:02:20] 
<-- SIP read from 192.168.1.165:50205: 
REGISTER sip:24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK264d402a
From: <sip:511@24.123.23.170;user=phone>
To: <sip:511@24.123.23.170;user=phone>
Call-ID: 000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165
CSeq: 11580 REGISTER
User-Agent: CSCO/7
Contact: <sip:511@192.168.1.165:5060>
Content-Length: 0
Expires: 60


[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK264d402a (58)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: From: <sip:511@24.123.23.170;user=phone> (40)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: To: <sip:511@24.123.23.170;user=phone> (38)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Call-ID: 000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165 (58)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: CSeq: 11580 REGISTER (20)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: User-Agent: CSCO/7 (18)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Contact: <sip:511@192.168.1.165:5060> (37)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Content-Length: 0 (17)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Expires: 60 (11)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10:  (0)

[Mar 10 12:02:20] --- (10 headers 0 lines) ---
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3168 sip_alloc: Allocating new SIP dialog for 000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165 - REGISTER (No RTP)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:02:20] Using latest REGISTER request as basis request

[Mar 10 12:02:20] Sending to 192.168.1.165 : 5060 (non-NAT)

[Mar 10 12:02:20] Transmitting (no NAT) to 192.168.1.165:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK264d402a;received=192.168.1.165
From: <sip:511@24.123.23.170;user=phone>
To: <sip:511@24.123.23.170;user=phone>
Call-ID: 000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165
CSeq: 11580 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:511@24.123.23.170>
Content-Length: 0


---

[Mar 10 12:02:20] Transmitting (no NAT) to 192.168.1.165:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK264d402a;received=192.168.1.165
From: <sip:511@24.123.23.170;user=phone>
To: <sip:511@24.123.23.170;user=phone>;tag=as1485df44
Call-ID: 000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165
CSeq: 11580 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="23ceef3e"
Content-Length: 0


---

[Mar 10 12:02:20] Scheduling destruction of call '000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165' in 15000 ms
[Mar 10 12:02:20] DEBUG[15137]: res_musiconhold.c:549 monmp3thread: Only wrote -1 of 640 bytes to pipe
[Mar 10 12:02:20] DEBUG[15137]: res_musiconhold.c:549 monmp3thread: Only wrote -1 of 640 bytes to pipe
[Mar 10 12:02:20] DEBUG[15137]: res_musiconhold.c:549 monmp3thread: Only wrote -1 of 640 bytes to pipe

[Mar 10 12:02:20] 
<-- SIP read from 192.168.1.165:50205: 
REGISTER sip:24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK43f67203
From: <sip:511@24.123.23.170;user=phone>
To: <sip:511@24.123.23.170;user=phone>
Call-ID: 000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165
CSeq: 11581 REGISTER
User-Agent: CSCO/7
Contact: <sip:511@192.168.1.165:5060>
Authorization: Digest username="511",realm="asterisk",uri="sip:24.123.23.170",response="dbc427749eeaf9a2df6b4e6c5ac3b878",nonce="23ceef3e",algorithm=MD5
Content-Length: 0
Expires: 60


[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK43f67203 (58)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: From: <sip:511@24.123.23.170;user=phone> (40)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: To: <sip:511@24.123.23.170;user=phone> (38)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Call-ID: 000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165 (58)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: CSeq: 11581 REGISTER (20)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: User-Agent: CSCO/7 (18)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Contact: <sip:511@192.168.1.165:5060> (37)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Authorization: Digest username="511",realm="asterisk",uri="sip:24.123.23.170",response="dbc427749eeaf9a2df6b4e6c5ac3b878",nonce="23ceef3e",algorithm=MD5 (152)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Content-Length: 0 (17)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Expires: 60 (11)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11:  (0)

[Mar 10 12:02:20] --- (11 headers 0 lines) ---
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:02:20] Using latest REGISTER request as basis request

[Mar 10 12:02:20] Sending to 192.168.1.165 : 5060 (non-NAT)

[Mar 10 12:02:20] Transmitting (no NAT) to 192.168.1.165:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK43f67203;received=192.168.1.165
From: <sip:511@24.123.23.170;user=phone>
To: <sip:511@24.123.23.170;user=phone>
Call-ID: 000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165
CSeq: 11581 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:511@24.123.23.170>
Content-Length: 0


---

[Mar 10 12:02:20] Transmitting (no NAT) to 192.168.1.165:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK43f67203;received=192.168.1.165
From: <sip:511@24.123.23.170;user=phone>
To: <sip:511@24.123.23.170;user=phone>;tag=as1485df44
Call-ID: 000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165
CSeq: 11581 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 60
Contact: <sip:511@192.168.1.165:5060>;expires=60
Date: Sat, 10 Mar 2007 17:02:20 GMT
Content-Length: 0


---

[Mar 10 12:02:20] Scheduling destruction of call '000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165' in 15000 ms
[Mar 10 12:02:20] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 511
[Mar 10 12:02:20] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/511 - state 1 (Not in use)
[Mar 10 12:02:20] DEBUG[15212]: app_queue.c:500 changethread: Device 'SIP/511' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:02:20] DEBUG[15137]: res_musiconhold.c:549 monmp3thread: Only wrote -1 of 640 bytes to pipe
[Mar 10 12:02:20] DEBUG[15137]: res_musiconhold.c:549 monmp3thread: Only wrote -1 of 640 bytes to pipe
[Mar 10 12:02:20] DEBUG[15137]: res_musiconhold.c:549 monmp3thread: Only wrote -1 of 640 bytes to pipe
[Mar 10 12:02:20] DEBUG[15137]: res_musiconhold.c:549 monmp3thread: Only wrote -1 of 640 bytes to pipe
[Mar 10 12:02:20] DEBUG[15137]: res_musiconhold.c:549 monmp3thread: Only wrote -1 of 640 bytes to pipe
[Mar 10 12:02:20] DEBUG[15137]: res_musiconhold.c:549 monmp3thread: Only wrote -1 of 640 bytes to pipe
[Mar 10 12:02:20] DEBUG[15137]: res_musiconhold.c:549 monmp3thread: Only wrote -1 of 640 bytes to pipe
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:1195 retrans_pkt: SIP TIMER: Rescheduling retransmission #133 (2) SIP/2.0 - 1
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:1209 retrans_pkt: ** SIP timers: Rescheduling retransmission 3 to 2000 ms (t1 500 ms (Retrans id #133)) 

[Mar 10 12:02:20] Retransmitting #2 (no NAT) to 192.168.1.85:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKaf8142956f618711e28805bf4c5289dfc;received=192.168.1.85
From: 523 <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f
To: <sip:593@24.123.23.170>;tag=as0da2dd59
Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170
CSeq: 388769 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:593@24.123.23.170>
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 15131 15134 IN IP4 192.168.1.83
s=session
c=IN IP4 192.168.1.83
t=0 0
m=audio 51000 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
[Mar 10 12:02:20] DEBUG[15137]: res_musiconhold.c:549 monmp3thread: Only wrote -1 of 640 bytes to pipe
[Mar 10 12:02:20] DEBUG[15137]: res_musiconhold.c:549 monmp3thread: Only wrote -1 of 640 bytes to pipe

[Mar 10 12:02:20] 
<-- SIP read from 192.168.1.85:5060: 
ACK sip:593@24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKg5a795cae7097e5e99fa3c05829cb73d8
CSeq: 388769 ACK
To: <sip:593@24.123.23.170>;tag=as0da2dd59
Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170
From: 523 <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f
User-Agent: Uniden SIP Phone p2 Ver BS4.77


[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: ACK sip:593@24.123.23.170 SIP/2.0 (33)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKg5a795cae7097e5e99fa3c05829cb73d8 (82)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: CSeq: 388769 ACK (16)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: To: <sip:593@24.123.23.170>;tag=as0da2dd59 (42)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170 (55)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: From: 523 <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f (74)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7:  (0)

[Mar 10 12:02:20] --- (7 headers 0 lines) ---
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received ACK (6) - Command in SIP ACK
[Mar 10 12:02:20] DEBUG[15137]: res_musiconhold.c:549 monmp3thread: Only wrote -1 of 640 bytes to pipe
[Mar 10 12:02:20] DEBUG[15137]: res_musiconhold.c:549 monmp3thread: Only wrote -1 of 640 bytes to pipe
[Mar 10 12:02:20] DEBUG[15137]: res_musiconhold.c:549 monmp3thread: Only wrote -1 of 640 bytes to pipe

[Mar 10 12:02:20] 
<-- SIP read from 192.168.1.97:5060: 
BYE sip:523@24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.97:5060;rport;branch=z9hG4bK2160127209
From: "Display Name" <sip:551@24.123.23.170>;tag=1341134544
To: <sip:523@24.123.23.170>;tag=as57536955
Call-ID: 3175191617@192.168.1.97
CSeq: 1002 BYE
max-forwards: 70
user-agent: UTSTARCOM F1000/Device ID-0007ba26174b
Content-Length: 0


[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: BYE sip:523@24.123.23.170 SIP/2.0 (33)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.97:5060;rport;branch=z9hG4bK2160127209 (65)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: From: "Display Name" <sip:551@24.123.23.170>;tag=1341134544 (59)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: To: <sip:523@24.123.23.170>;tag=as57536955 (42)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Call-ID: 3175191617@192.168.1.97 (32)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: CSeq: 1002 BYE (14)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: max-forwards: 70 (16)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: user-agent: UTSTARCOM F1000/Device ID-0007ba26174b (50)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Content-Length: 0 (17)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9:  (0)

[Mar 10 12:02:20] --- (9 headers 0 lines) ---
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received BYE (8) - Command in SIP BYE

[Mar 10 12:02:20] Sending to 192.168.1.97 : 5060 (NAT)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:10876 handle_request_bye: Received bye, issuing owner hangup
.
[Mar 10 12:02:20] Transmitting (NAT) to 192.168.1.97:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.97:5060;branch=z9hG4bK2160127209;received=192.168.1.97;rport=5060
From: "Display Name" <sip:551@24.123.23.170>;tag=1341134544
To: <sip:523@24.123.23.170>;tag=as57536955
Call-ID: 3175191617@192.168.1.97
CSeq: 1002 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:523@24.123.23.170>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


---
[Mar 10 12:02:20] DEBUG[15201]: channel.c:2409 set_format: Set channel SIP/551-08518cd8 to write format ulaw

[Mar 10 12:02:20]     -- Stopped music on hold on SIP/551-08518cd8
[Mar 10 12:02:20] DEBUG[15201]: channel.c:1761 ast_settimeout: Scheduling timer at 0 sample intervals
[Mar 10 12:02:20] DEBUG[15201]: rtp.c:1718 ast_rtp_bridge: Oooh, got a hangup
[Mar 10 12:02:20] DEBUG[15201]: chan_sip.c:13028 sip_set_rtp_peer: Sending reinvite on SIP '3890f17d116db7343bb188d42c093fd2@24.123.23.170' - It's audio soon redirected to IP 24.123.23.170

[Mar 10 12:02:20] set_destination: Parsing <sip:523@192.168.1.85:5060> for address/port to send to

[Mar 10 12:02:20] set_destination: set destination to 192.168.1.85, port 5060

[Mar 10 12:02:20] We're at 24.123.23.170 port 14572

[Mar 10 12:02:20] Video is at 24.123.23.170 port 16640

[Mar 10 12:02:20] Adding codec 0x4 (ulaw) to SDP

[Mar 10 12:02:20] Adding non-codec 0x1 (telephone-event) to SDP
[Mar 10 12:02:20] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 0: INVITE sip:523@192.168.1.85:5060 SIP/2.0 (40)
[Mar 10 12:02:20] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK33bb110c (58)
[Mar 10 12:02:20] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 2: From: <sip:551@24.123.23.170>;tag=as6fedd0f9 (44)
[Mar 10 12:02:20] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 3: To: 523 <sip:523@192.168.1.85:5060>;tag=a997c53df28f4a860026417db9e2f2ab (72)
[Mar 10 12:02:20] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 4: Contact: <sip:551@24.123.23.170> (32)
[Mar 10 12:02:20] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 5: Call-ID: 3890f17d116db7343bb188d42c093fd2@24.123.23.170 (55)
[Mar 10 12:02:20] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 6: CSeq: 104 INVITE (16)
[Mar 10 12:02:20] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 7: User-Agent: Asterisk PBX (24)
[Mar 10 12:02:20] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 8: Max-Forwards: 70 (16)
[Mar 10 12:02:20] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66)
[Mar 10 12:02:20] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 10: X-asterisk-info: SIP re-invite (RTP bridge) (43)
[Mar 10 12:02:20] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 11: Content-Type: application/sdp (29)
[Mar 10 12:02:20] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 12: Content-Length: 218 (19)
[Mar 10 12:02:20] DEBUG[15201]: chan_sip.c:3392 parse_request: Header 13:  (0)
[Mar 10 12:02:20] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: v=0 (3)
[Mar 10 12:02:20] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: o=root 15131 15134 IN IP4 24.123.23.170 (39)
[Mar 10 12:02:20] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: s=session (9)
[Mar 10 12:02:20] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: c=IN IP4 24.123.23.170 (22)
[Mar 10 12:02:20] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: t=0 0 (5)
[Mar 10 12:02:20] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: m=audio 14572 RTP/AVP 0 101 (27)
[Mar 10 12:02:20] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: a=rtpmap:0 PCMU/8000 (20)
[Mar 10 12:02:20] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33)
[Mar 10 12:02:20] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: a=fmtp:101 0-16 (15)
[Mar 10 12:02:20] DEBUG[15201]: chan_sip.c:3424 parse_request: Line: a=silenceSupp:off - - - - (25)

[Mar 10 12:02:20] 13 headers, 10 lines

[Mar 10 12:02:20] Reliably Transmitting (no NAT) to 192.168.1.85:5060:
INVITE sip:523@192.168.1.85:5060 SIP/2.0
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK33bb110c
From: <sip:551@24.123.23.170>;tag=as6fedd0f9
To: 523 <sip:523@192.168.1.85:5060>;tag=a997c53df28f4a860026417db9e2f2ab
Contact: <sip:551@24.123.23.170>
Call-ID: 3890f17d116db7343bb188d42c093fd2@24.123.23.170
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 218

v=0
o=root 15131 15134 IN IP4 24.123.23.170
s=session
c=IN IP4 24.123.23.170
t=0 0
m=audio 14572 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
[Mar 10 12:02:20] DEBUG[15201]: chan_sip.c:1304 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id  #145
[Mar 10 12:02:20] DEBUG[15201]: channel.c:3600 ast_channel_bridge: Returning from native bridge, channels: SIP/551-08518cd8, SIP/523-08521b58
[Mar 10 12:02:20] DEBUG[15201]: channel.c:1373 ast_hangup: Hanging up channel 'SIP/523-08521b58'
[Mar 10 12:02:20] DEBUG[15201]: chan_sip.c:2427 sip_hangup: Hangup call SIP/523-08521b58, SIP callid 3890f17d116db7343bb188d42c093fd2@24.123.23.170)
[Mar 10 12:02:20] DEBUG[15201]: chan_sip.c:2435 sip_hangup: update_call_counter(523) - decrement call limit counter
[Mar 10 12:02:20] DEBUG[15201]: chan_sip.c:2217 update_call_counter: Updating call counter for outgoing call

[Mar 10 12:02:20] Scheduling destruction of call '3890f17d116db7343bb188d42c093fd2@24.123.23.170' in 32000 ms
[Mar 10 12:02:20] DEBUG[15201]: app_dial.c:1635 dial_exec_full: Exiting with DIALSTATUS=ANSWER.
[Mar 10 12:02:20] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 523
[Mar 10 12:02:20] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/523 - state 2 (In use)
[Mar 10 12:02:20] DEBUG[15201]: pbx.c:2316 __ast_pbx_run: Spawn extension (smvoice-sip,523,6) exited non-zero on 'SIP/551-08518cd8'

[Mar 10 12:02:20]   == Spawn extension (smvoice-sip, 523, 6) exited non-zero on 'SIP/551-08518cd8'
[Mar 10 12:02:20] DEBUG[15201]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '"551 551" <551>'
[Mar 10 12:02:20] DEBUG[15201]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '551'
[Mar 10 12:02:20] DEBUG[15201]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '523'
[Mar 10 12:02:20] DEBUG[15201]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'smvoice-sip'
[Mar 10 12:02:20] DEBUG[15201]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'SIP/551-08518cd8'
[Mar 10 12:02:20] DEBUG[15201]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'SIP/523-08521b58'
[Mar 10 12:02:20] DEBUG[15201]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'Dial'
[Mar 10 12:02:20] DEBUG[15201]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'SIP/523|20|'
[Mar 10 12:02:20] DEBUG[15201]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '2007-03-10 12:02:11'
[Mar 10 12:02:20] DEBUG[15201]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '2007-03-10 12:02:12'
[Mar 10 12:02:20] DEBUG[15201]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '2007-03-10 12:02:20'
[Mar 10 12:02:20] DEBUG[15201]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '9'
[Mar 10 12:02:20] DEBUG[15201]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '8'
[Mar 10 12:02:20] DEBUG[15201]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'ANSWERED'
[Mar 10 12:02:20] DEBUG[15201]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION'
[Mar 10 12:02:20] DEBUG[15201]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '(null)'
[Mar 10 12:02:20] DEBUG[15201]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '1173546131.2'
[Mar 10 12:02:20] DEBUG[15201]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '(null)'
[Mar 10 12:02:20] DEBUG[15201]: channel.c:1373 ast_hangup: Hanging up channel 'SIP/551-08518cd8'
[Mar 10 12:02:20] DEBUG[15201]: chan_sip.c:2427 sip_hangup: Hangup call SIP/551-08518cd8, SIP callid 3175191617@192.168.1.97)
[Mar 10 12:02:20] DEBUG[15201]: chan_sip.c:2435 sip_hangup: update_call_counter(551) - decrement call limit counter
[Mar 10 12:02:20] DEBUG[15201]: chan_sip.c:2217 update_call_counter: Updating call counter for outgoing call
[Mar 10 12:02:20] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 551
[Mar 10 12:02:20] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/551 - state 1 (Not in use)
[Mar 10 12:02:20] DEBUG[15214]: app_queue.c:500 changethread: Device 'SIP/551' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:02:20] DEBUG[15213]: app_queue.c:500 changethread: Device 'SIP/523' changed to state '2' (In use) but we don't care because they're not a member of any queue.

[Mar 10 12:02:20] 
<-- SIP read from 192.168.1.85:5060: 
SIP/2.0 200 OK
To: 523 <sip:523@192.168.1.85:5060>;tag=a997c53df28f4a860026417db9e2f2ab
Contact: <sip:523@192.168.1.85:5060>
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK33bb110c
From: <sip:551@24.123.23.170>;tag=as6fedd0f9
Call-ID: 3890f17d116db7343bb188d42c093fd2@24.123.23.170
CSeq: 104 INVITE
Content-Length: 218
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK

v=0
o=- 15131 298890 IN IP4 192.168.1.85
s=session
c=IN IP4 0.0.0.0
t=0 0
m=audio 30014 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,16,17,18
a=silenceSupp:off - - - -

[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: SIP/2.0 200 OK (14)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: To: 523 <sip:523@192.168.1.85:5060>;tag=a997c53df28f4a860026417db9e2f2ab (72)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: Contact: <sip:523@192.168.1.85:5060> (36)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK33bb110c (58)
[Mar 10 12:02:20] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: From: <sip:551@24.123.23.170>;tag=as6fedd0f9 (44)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: Call-ID: 3890f17d116db7343bb188d42c093fd2@24.123.23.170 (55)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: CSeq: 104 INVITE (16)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Content-Length: 218 (19)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Supported: sip-cc, sip-cc-01, replaces, timer (45)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10:  (0)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: v=0 (3)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: o=- 15131 298890 IN IP4 192.168.1.85 (36)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: s=session (9)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: c=IN IP4 0.0.0.0 (16)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: t=0 0 (5)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: m=audio 30014 RTP/AVP 0 101 (27)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:0 PCMU/8000 (20)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=fmtp:101 0-15,16,17,18 (24)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=silenceSupp:off - - - - (25)

[Mar 10 12:02:22] --- (10 headers 10 lines) ---
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:1390 __sip_ack: Acked pending invite 104
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:1401 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #145
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:1412 __sip_ack: Stopping retransmission on '3890f17d116db7343bb188d42c093fd2@24.123.23.170' of Request 104: Match Found
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:9642 handle_response_invite: SIP response 200 to standard invite
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:6137 build_route: build_route: Retaining previous route: <sip:523@192.168.1.85:5060>

[Mar 10 12:02:22] set_destination: Parsing <sip:523@192.168.1.85:5060> for address/port to send to

[Mar 10 12:02:22] set_destination: set destination to 192.168.1.85, port 5060

[Mar 10 12:02:22] Transmitting (no NAT) to 192.168.1.85:5060:
ACK sip:523@192.168.1.85:5060 SIP/2.0
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK506226df
From: <sip:551@24.123.23.170>;tag=as6fedd0f9
To: 523 <sip:523@192.168.1.85:5060>;tag=a997c53df28f4a860026417db9e2f2ab
Contact: <sip:551@24.123.23.170>
Call-ID: 3890f17d116db7343bb188d42c093fd2@24.123.23.170
CSeq: 104 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---

[Mar 10 12:02:22] set_destination: Parsing <sip:523@192.168.1.85:5060> for address/port to send to

[Mar 10 12:02:22] set_destination: set destination to 192.168.1.85, port 5060

[Mar 10 12:02:22] Reliably Transmitting (no NAT) to 192.168.1.85:5060:
BYE sip:523@192.168.1.85:5060 SIP/2.0
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK7286f5b5
From: <sip:551@24.123.23.170>;tag=as6fedd0f9
To: 523 <sip:523@192.168.1.85:5060>;tag=a997c53df28f4a860026417db9e2f2ab
Contact: <sip:551@24.123.23.170>
Call-ID: 3890f17d116db7343bb188d42c093fd2@24.123.23.170
CSeq: 105 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:1304 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id  #147

[Mar 10 12:02:22] Scheduling destruction of call '3890f17d116db7343bb188d42c093fd2@24.123.23.170' in 32000 ms
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:1195 retrans_pkt: SIP TIMER: Rescheduling retransmission #133 (3) SIP/2.0 - 1
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:1209 retrans_pkt: ** SIP timers: Rescheduling retransmission 4 to 4000 ms (t1 500 ms (Retrans id #133)) 

[Mar 10 12:02:22] Retransmitting #3 (no NAT) to 192.168.1.85:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKaf8142956f618711e28805bf4c5289dfc;received=192.168.1.85
From: 523 <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f
To: <sip:593@24.123.23.170>;tag=as0da2dd59
Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170
CSeq: 388769 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:593@24.123.23.170>
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 15131 15134 IN IP4 192.168.1.83
s=session
c=IN IP4 192.168.1.83
t=0 0
m=audio 51000 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---

[Mar 10 12:02:22] Destroying call '3175191617@192.168.1.97'

[Mar 10 12:02:22] 
<-- SIP read from 68.58.36.157:5060: 


[Mar 10 12:02:22] --- (0 headers 0 lines) Nat keepalive ---

[Mar 10 12:02:22] 
<-- SIP read from 192.168.1.85:5060: 
SIP/2.0 200 OK
To: 523 <sip:523@192.168.1.85:5060>;tag=a997c53df28f4a860026417db9e2f2ab
Contact: <sip:523@192.168.1.85:5060>
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK33bb110c
From: <sip:551@24.123.23.170>;tag=as6fedd0f9
Call-ID: 3890f17d116db7343bb188d42c093fd2@24.123.23.170
CSeq: 104 INVITE
Content-Length: 218
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK

v=0
o=- 15131 298890 IN IP4 192.168.1.85
s=session
c=IN IP4 0.0.0.0
t=0 0
m=audio 30014 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,16,17,18
a=silenceSupp:off - - - -

[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: SIP/2.0 200 OK (14)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: To: 523 <sip:523@192.168.1.85:5060>;tag=a997c53df28f4a860026417db9e2f2ab (72)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: Contact: <sip:523@192.168.1.85:5060> (36)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK33bb110c (58)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: From: <sip:551@24.123.23.170>;tag=as6fedd0f9 (44)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: Call-ID: 3890f17d116db7343bb188d42c093fd2@24.123.23.170 (55)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: CSeq: 104 INVITE (16)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Content-Length: 218 (19)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Supported: sip-cc, sip-cc-01, replaces, timer (45)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10:  (0)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: v=0 (3)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: o=- 15131 298890 IN IP4 192.168.1.85 (36)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: s=session (9)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: c=IN IP4 0.0.0.0 (16)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: t=0 0 (5)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: m=audio 30014 RTP/AVP 0 101 (27)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:0 PCMU/8000 (20)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=fmtp:101 0-15,16,17,18 (24)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=silenceSupp:off - - - - (25)

[Mar 10 12:02:22] --- (10 headers 10 lines) ---
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:1412 __sip_ack: Stopping retransmission on '3890f17d116db7343bb188d42c093fd2@24.123.23.170' of Request 104: Match Not Found
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:9642 handle_response_invite: SIP response 200 to standard invite
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:6137 build_route: build_route: Retaining previous route: <sip:523@192.168.1.85:5060>

[Mar 10 12:02:22] set_destination: Parsing <sip:523@192.168.1.85:5060> for address/port to send to

[Mar 10 12:02:22] set_destination: set destination to 192.168.1.85, port 5060

[Mar 10 12:02:22] Transmitting (no NAT) to 192.168.1.85:5060:
ACK sip:523@192.168.1.85:5060 SIP/2.0
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK04c44ef4
From: <sip:551@24.123.23.170>;tag=as6fedd0f9
To: 523 <sip:523@192.168.1.85:5060>;tag=a997c53df28f4a860026417db9e2f2ab
Contact: <sip:551@24.123.23.170>
Call-ID: 3890f17d116db7343bb188d42c093fd2@24.123.23.170
CSeq: 104 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---

[Mar 10 12:02:22] 
<-- SIP read from 192.168.1.86:5062: 


[Mar 10 12:02:22] --- (0 headers 0 lines) Nat keepalive ---

[Mar 10 12:02:22] 
<-- SIP read from 192.168.1.165:50204: 
REGISTER sip:24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK1978d846
From: <sip:510@24.123.23.170;user=phone>
To: <sip:510@24.123.23.170;user=phone>
Call-ID: 000ff78d-ebb20002-23a7f663-5bd85286@192.168.1.165
CSeq: 11593 REGISTER
User-Agent: CSCO/7
Contact: <sip:510@192.168.1.165:5060>
Content-Length: 0
Expires: 60


[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK1978d846 (58)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: From: <sip:510@24.123.23.170;user=phone> (40)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: To: <sip:510@24.123.23.170;user=phone> (38)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Call-ID: 000ff78d-ebb20002-23a7f663-5bd85286@192.168.1.165 (58)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: CSeq: 11593 REGISTER (20)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: User-Agent: CSCO/7 (18)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Contact: <sip:510@192.168.1.165:5060> (37)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Content-Length: 0 (17)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Expires: 60 (11)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10:  (0)

[Mar 10 12:02:22] --- (10 headers 0 lines) ---
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3168 sip_alloc: Allocating new SIP dialog for 000ff78d-ebb20002-23a7f663-5bd85286@192.168.1.165 - REGISTER (No RTP)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:02:22] Using latest REGISTER request as basis request

[Mar 10 12:02:22] Sending to 192.168.1.165 : 5060 (non-NAT)

[Mar 10 12:02:22] Transmitting (no NAT) to 192.168.1.165:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK1978d846;received=192.168.1.165
From: <sip:510@24.123.23.170;user=phone>
To: <sip:510@24.123.23.170;user=phone>
Call-ID: 000ff78d-ebb20002-23a7f663-5bd85286@192.168.1.165
CSeq: 11593 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:510@24.123.23.170>
Content-Length: 0


---

[Mar 10 12:02:22] Transmitting (no NAT) to 192.168.1.165:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK1978d846;received=192.168.1.165
From: <sip:510@24.123.23.170;user=phone>
To: <sip:510@24.123.23.170;user=phone>;tag=as05a23dd7
Call-ID: 000ff78d-ebb20002-23a7f663-5bd85286@192.168.1.165
CSeq: 11593 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="23bde679"
Content-Length: 0


---

[Mar 10 12:02:22] Scheduling destruction of call '000ff78d-ebb20002-23a7f663-5bd85286@192.168.1.165' in 15000 ms

[Mar 10 12:02:22] 
<-- SIP read from 192.168.1.85:5060: 
SIP/2.0 200 OK
To: 523 <sip:523@192.168.1.85:5060>;tag=a997c53df28f4a860026417db9e2f2ab
Contact: <sip:523@192.168.1.85:5060>
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK33bb110c
From: <sip:551@24.123.23.170>;tag=as6fedd0f9
Call-ID: 3890f17d116db7343bb188d42c093fd2@24.123.23.170
CSeq: 104 INVITE
Content-Length: 218
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK

v=0
o=- 15131 298890 IN IP4 192.168.1.85
s=session
c=IN IP4 0.0.0.0
t=0 0
m=audio 30014 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,16,17,18
a=silenceSupp:off - - - -

[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: SIP/2.0 200 OK (14)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: To: 523 <sip:523@192.168.1.85:5060>;tag=a997c53df28f4a860026417db9e2f2ab (72)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: Contact: <sip:523@192.168.1.85:5060> (36)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK33bb110c (58)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: From: <sip:551@24.123.23.170>;tag=as6fedd0f9 (44)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: Call-ID: 3890f17d116db7343bb188d42c093fd2@24.123.23.170 (55)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: CSeq: 104 INVITE (16)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Content-Length: 218 (19)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Supported: sip-cc, sip-cc-01, replaces, timer (45)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10:  (0)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: v=0 (3)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: o=- 15131 298890 IN IP4 192.168.1.85 (36)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: s=session (9)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: c=IN IP4 0.0.0.0 (16)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: t=0 0 (5)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: m=audio 30014 RTP/AVP 0 101 (27)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:0 PCMU/8000 (20)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=fmtp:101 0-15,16,17,18 (24)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=silenceSupp:off - - - - (25)

[Mar 10 12:02:22] --- (10 headers 10 lines) ---
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:1412 __sip_ack: Stopping retransmission on '3890f17d116db7343bb188d42c093fd2@24.123.23.170' of Request 104: Match Not Found
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:9642 handle_response_invite: SIP response 200 to standard invite
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:6137 build_route: build_route: Retaining previous route: <sip:523@192.168.1.85:5060>

[Mar 10 12:02:22] set_destination: Parsing <sip:523@192.168.1.85:5060> for address/port to send to

[Mar 10 12:02:22] set_destination: set destination to 192.168.1.85, port 5060

[Mar 10 12:02:22] Transmitting (no NAT) to 192.168.1.85:5060:
ACK sip:523@192.168.1.85:5060 SIP/2.0
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK41e2a0ba
From: <sip:551@24.123.23.170>;tag=as6fedd0f9
To: 523 <sip:523@192.168.1.85:5060>;tag=a997c53df28f4a860026417db9e2f2ab
Contact: <sip:551@24.123.23.170>
Call-ID: 3890f17d116db7343bb188d42c093fd2@24.123.23.170
CSeq: 104 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---

[Mar 10 12:02:22] 
<-- SIP read from 192.168.1.165:50204: 
REGISTER sip:24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK1978d846
From: <sip:510@24.123.23.170;user=phone>
To: <sip:510@24.123.23.170;user=phone>
Call-ID: 000ff78d-ebb20002-23a7f663-5bd85286@192.168.1.165
CSeq: 11593 REGISTER
User-Agent: CSCO/7
Contact: <sip:510@192.168.1.165:5060>
Content-Length: 0
Expires: 60


[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK1978d846 (58)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: From: <sip:510@24.123.23.170;user=phone> (40)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: To: <sip:510@24.123.23.170;user=phone> (38)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Call-ID: 000ff78d-ebb20002-23a7f663-5bd85286@192.168.1.165 (58)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: CSeq: 11593 REGISTER (20)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: User-Agent: CSCO/7 (18)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Contact: <sip:510@192.168.1.165:5060> (37)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Content-Length: 0 (17)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Expires: 60 (11)
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10:  (0)

[Mar 10 12:02:22] --- (10 headers 0 lines) ---
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER
[Mar 10 12:02:22] DEBUG[15149]: chan_sip.c:11239 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 11593, ours 11593)

[Mar 10 12:02:22] Using latest REGISTER request as basis request

[Mar 10 12:02:22] Sending to 192.168.1.165 : 5060 (non-NAT)

[Mar 10 12:02:22] Transmitting (no NAT) to 192.168.1.165:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK1978d846;received=192.168.1.165
From: <sip:510@24.123.23.170;user=phone>
To: <sip:510@24.123.23.170;user=phone>
Call-ID: 000ff78d-ebb20002-23a7f663-5bd85286@192.168.1.165
CSeq: 11593 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:510@24.123.23.170>
Content-Length: 0


---

[Mar 10 12:02:22] Transmitting (no NAT) to 192.168.1.165:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK1978d846;received=192.168.1.165
From: <sip:510@24.123.23.170;user=phone>
To: <sip:510@24.123.23.170;user=phone>;tag=as05a23dd7
Call-ID: 000ff78d-ebb20002-23a7f663-5bd85286@192.168.1.165
CSeq: 11593 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="23bde679"
Content-Length: 0


---

[Mar 10 12:02:22] Scheduling destruction of call '000ff78d-ebb20002-23a7f663-5bd85286@192.168.1.165' in 15000 ms

[Mar 10 12:02:23] 
<-- SIP read from 192.168.1.165:50204: 
REGISTER sip:24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK0a4c6021
From: <sip:510@24.123.23.170;user=phone>
To: <sip:510@24.123.23.170;user=phone>
Call-ID: 000ff78d-ebb20002-23a7f663-5bd85286@192.168.1.165
CSeq: 11594 REGISTER
User-Agent: CSCO/7
Contact: <sip:510@192.168.1.165:5060>
Authorization: Digest username="510",realm="asterisk",uri="sip:24.123.23.170",response="f2c25ee753487c4504c6446cab854da3",nonce="23bde679",algorithm=MD5
Content-Length: 0
Expires: 60


[Mar 10 12:02:23] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34)
[Mar 10 12:02:23] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK0a4c6021 (58)
[Mar 10 12:02:23] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: From: <sip:510@24.123.23.170;user=phone> (40)
[Mar 10 12:02:23] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: To: <sip:510@24.123.23.170;user=phone> (38)
[Mar 10 12:02:23] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Call-ID: 000ff78d-ebb20002-23a7f663-5bd85286@192.168.1.165 (58)
[Mar 10 12:02:23] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: CSeq: 11594 REGISTER (20)
[Mar 10 12:02:23] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: User-Agent: CSCO/7 (18)
[Mar 10 12:02:23] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Contact: <sip:510@192.168.1.165:5060> (37)
[Mar 10 12:02:23] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Authorization: Digest username="510",realm="asterisk",uri="sip:24.123.23.170",response="f2c25ee753487c4504c6446cab854da3",nonce="23bde679",algorithm=MD5 (152)
[Mar 10 12:02:23] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Content-Length: 0 (17)
[Mar 10 12:02:23] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Expires: 60 (11)
[Mar 10 12:02:23] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11:  (0)

[Mar 10 12:02:23] --- (11 headers 0 lines) ---
[Mar 10 12:02:23] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:02:23] Using latest REGISTER request as basis request

[Mar 10 12:02:23] Sending to 192.168.1.165 : 5060 (non-NAT)

[Mar 10 12:02:23] Transmitting (no NAT) to 192.168.1.165:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK0a4c6021;received=192.168.1.165
From: <sip:510@24.123.23.170;user=phone>
To: <sip:510@24.123.23.170;user=phone>
Call-ID: 000ff78d-ebb20002-23a7f663-5bd85286@192.168.1.165
CSeq: 11594 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:510@24.123.23.170>
Content-Length: 0


---

[Mar 10 12:02:23] Transmitting (no NAT) to 192.168.1.165:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK0a4c6021;received=192.168.1.165
From: <sip:510@24.123.23.170;user=phone>
To: <sip:510@24.123.23.170;user=phone>;tag=as05a23dd7
Call-ID: 000ff78d-ebb20002-23a7f663-5bd85286@192.168.1.165
CSeq: 11594 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 60
Contact: <sip:510@192.168.1.165:5060>;expires=60
Date: Sat, 10 Mar 2007 17:02:23 GMT
Content-Length: 0


---

[Mar 10 12:02:23] Scheduling destruction of call '000ff78d-ebb20002-23a7f663-5bd85286@192.168.1.165' in 15000 ms
[Mar 10 12:02:23] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 510
[Mar 10 12:02:23] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/510 - state 1 (Not in use)
[Mar 10 12:02:23] DEBUG[15215]: app_queue.c:500 changethread: Device 'SIP/510' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.

[Mar 10 12:02:23] 
<-- SIP read from 192.168.1.85:5060: 
SIP/2.0 200 OK
To: 523 <sip:523@192.168.1.85:5060>;tag=a997c53df28f4a860026417db9e2f2ab
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK7286f5b5
From: <sip:551@24.123.23.170>;tag=as6fedd0f9
Call-ID: 3890f17d116db7343bb188d42c093fd2@24.123.23.170
CSeq: 105 BYE
Content-Length: 0
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK


[Mar 10 12:02:23] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: SIP/2.0 200 OK (14)
[Mar 10 12:02:23] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: To: 523 <sip:523@192.168.1.85:5060>;tag=a997c53df28f4a860026417db9e2f2ab (72)
[Mar 10 12:02:23] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK7286f5b5 (58)
[Mar 10 12:02:23] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: From: <sip:551@24.123.23.170>;tag=as6fedd0f9 (44)
[Mar 10 12:02:23] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Call-ID: 3890f17d116db7343bb188d42c093fd2@24.123.23.170 (55)
[Mar 10 12:02:23] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: CSeq: 105 BYE (13)
[Mar 10 12:02:23] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: Content-Length: 0 (17)
[Mar 10 12:02:23] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Supported: sip-cc, sip-cc-01, replaces, timer (45)
[Mar 10 12:02:23] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62)
[Mar 10 12:02:23] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9:  (0)

[Mar 10 12:02:23] --- (9 headers 0 lines) ---
[Mar 10 12:02:23] DEBUG[15149]: chan_sip.c:1401 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #147
[Mar 10 12:02:23] DEBUG[15149]: chan_sip.c:1412 __sip_ack: Stopping retransmission on '3890f17d116db7343bb188d42c093fd2@24.123.23.170' of Request 105: Match Found

[Mar 10 12:02:23] Destroying call '3890f17d116db7343bb188d42c093fd2@24.123.23.170'
[Mar 10 12:02:26] DEBUG[15149]: chan_sip.c:1195 retrans_pkt: SIP TIMER: Rescheduling retransmission #133 (4) SIP/2.0 - 1
[Mar 10 12:02:26] DEBUG[15149]: chan_sip.c:1209 retrans_pkt: ** SIP timers: Rescheduling retransmission 5 to 4000 ms (t1 500 ms (Retrans id #133)) 

[Mar 10 12:02:26] Retransmitting #4 (no NAT) to 192.168.1.85:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKaf8142956f618711e28805bf4c5289dfc;received=192.168.1.85
From: 523 <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f
To: <sip:593@24.123.23.170>;tag=as0da2dd59
Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170
CSeq: 388769 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:593@24.123.23.170>
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 15131 15134 IN IP4 192.168.1.83
s=session
c=IN IP4 192.168.1.83
t=0 0
m=audio 51000 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---

[Mar 10 12:02:27] 
<-- SIP read from 192.168.1.97:5060: 


[Mar 10 12:02:27] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0:  (0)

[Mar 10 12:02:27] --- (0 headers 0 lines) Nat keepalive ---

[Mar 10 12:02:29] 
<-- SIP read from 192.168.1.76:5060: 
REGISTER sip:24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bKnbb2bde880a185b5552177cd5e1abd46a
Call-ID: 55ae66476793@24.123.23.170
CSeq: 38999 REGISTER
From: <sip:525@24.123.23.170>;tag=0e928b1b018f140ea648cfb9cd199547
To: <sip:525@24.123.23.170>
Contact: <sip:525@192.168.1.76:5060>;expires=60;q=0.50
User-Agent: Uniden SIP Phone p2 Ver BS4.77
Max-Forwards: 70
Content-Length: 0
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK
Authorization: Digest realm="asterisk",   nonce="1c1fbe40",   algorithm=MD5,   uri="sip:24.123.23.170",   username="525",   response="c0a61b13003ee095b1dbbf89f43d7f60"


[Mar 10 12:02:29] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34)
[Mar 10 12:02:29] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bKnbb2bde880a185b5552177cd5e1abd46a (82)
[Mar 10 12:02:29] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: Call-ID: 55ae66476793@24.123.23.170 (35)
[Mar 10 12:02:29] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: CSeq: 38999 REGISTER (20)
[Mar 10 12:02:29] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: From: <sip:525@24.123.23.170>;tag=0e928b1b018f140ea648cfb9cd199547 (66)
[Mar 10 12:02:29] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: To: <sip:525@24.123.23.170> (27)
[Mar 10 12:02:29] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: Contact: <sip:525@192.168.1.76:5060>;expires=60;q=0.50 (54)
[Mar 10 12:02:29] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42)
[Mar 10 12:02:29] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Max-Forwards: 70 (16)
[Mar 10 12:02:29] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Content-Length: 0 (17)
[Mar 10 12:02:29] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45)
[Mar 10 12:02:29] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62)
[Mar 10 12:02:29] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 12: Authorization: Digest realm="asterisk",   nonce="1c1fbe40",   algorithm=MD5,   uri="sip:24.123.23.170",   username="525",   response="c0a61b13003ee095b1dbbf89f43d7f60" (167)
[Mar 10 12:02:29] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 13:  (0)

[Mar 10 12:02:29] --- (13 headers 0 lines) ---
[Mar 10 12:02:29] DEBUG[15149]: chan_sip.c:3168 sip_alloc: Allocating new SIP dialog for 55ae66476793@24.123.23.170 - REGISTER (No RTP)
[Mar 10 12:02:29] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:02:29] Using latest REGISTER request as basis request

[Mar 10 12:02:29] Sending to 192.168.1.76 : 5060 (non-NAT)

[Mar 10 12:02:29] Transmitting (no NAT) to 192.168.1.76:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bKnbb2bde880a185b5552177cd5e1abd46a;received=192.168.1.76
From: <sip:525@24.123.23.170>;tag=0e928b1b018f140ea648cfb9cd199547
To: <sip:525@24.123.23.170>
Call-ID: 55ae66476793@24.123.23.170
CSeq: 38999 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:525@24.123.23.170>
Content-Length: 0


---

[Mar 10 12:02:29] Transmitting (no NAT) to 192.168.1.76:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bKnbb2bde880a185b5552177cd5e1abd46a;received=192.168.1.76
From: <sip:525@24.123.23.170>;tag=0e928b1b018f140ea648cfb9cd199547
To: <sip:525@24.123.23.170>;tag=as4477b37f
Call-ID: 55ae66476793@24.123.23.170
CSeq: 38999 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7cb02f8e"
Content-Length: 0


---

[Mar 10 12:02:29] Scheduling destruction of call '55ae66476793@24.123.23.170' in 15000 ms
[Mar 10 12:02:29] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 526
[Mar 10 12:02:29] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/526 - state 5 (Unavailable)
[Mar 10 12:02:29] DEBUG[15216]: app_queue.c:500 changethread: Device 'SIP/526' changed to state '5' (Unavailable) but we don't care because they're not a member of any queue.

[Mar 10 12:02:29] 
<-- SIP read from 192.168.1.76:5060: 
REGISTER sip:24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bKf8ba08c3cb5e99023844f3ffce127d15e
CSeq: 39000 REGISTER
Call-ID: 55ae66476793@24.123.23.170
From: <sip:525@24.123.23.170>;tag=0e928b1b018f140ea648cfb9cd199547
To: <sip:525@24.123.23.170>
Contact: <sip:525@192.168.1.76:5060>;expires=60;q=0.50
User-Agent: Uniden SIP Phone p2 Ver BS4.77
Max-Forwards: 70
Content-Length: 0
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK
Authorization: Digest realm="asterisk",   nonce="7cb02f8e",   algorithm=MD5,   uri="sip:24.123.23.170",   username="525",   response="b0ff77ebfb3396716ffe9d95d5ee7d06"


[Mar 10 12:02:29] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34)
[Mar 10 12:02:29] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bKf8ba08c3cb5e99023844f3ffce127d15e (82)
[Mar 10 12:02:29] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: CSeq: 39000 REGISTER (20)
[Mar 10 12:02:29] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: Call-ID: 55ae66476793@24.123.23.170 (35)
[Mar 10 12:02:29] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: From: <sip:525@24.123.23.170>;tag=0e928b1b018f140ea648cfb9cd199547 (66)
[Mar 10 12:02:29] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: To: <sip:525@24.123.23.170> (27)
[Mar 10 12:02:29] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: Contact: <sip:525@192.168.1.76:5060>;expires=60;q=0.50 (54)
[Mar 10 12:02:29] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42)
[Mar 10 12:02:29] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Max-Forwards: 70 (16)
[Mar 10 12:02:29] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Content-Length: 0 (17)
[Mar 10 12:02:29] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45)
[Mar 10 12:02:29] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62)
[Mar 10 12:02:29] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 12: Authorization: Digest realm="asterisk",   nonce="7cb02f8e",   algorithm=MD5,   uri="sip:24.123.23.170",   username="525",   response="b0ff77ebfb3396716ffe9d95d5ee7d06" (167)
[Mar 10 12:02:29] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 13:  (0)

[Mar 10 12:02:29] --- (13 headers 0 lines) ---
[Mar 10 12:02:29] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:02:29] Using latest REGISTER request as basis request

[Mar 10 12:02:29] Sending to 192.168.1.76 : 5060 (non-NAT)

[Mar 10 12:02:29] Transmitting (no NAT) to 192.168.1.76:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bKf8ba08c3cb5e99023844f3ffce127d15e;received=192.168.1.76
From: <sip:525@24.123.23.170>;tag=0e928b1b018f140ea648cfb9cd199547
To: <sip:525@24.123.23.170>
Call-ID: 55ae66476793@24.123.23.170
CSeq: 39000 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:525@24.123.23.170>
Content-Length: 0


---

[Mar 10 12:02:29] Transmitting (no NAT) to 192.168.1.76:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bKf8ba08c3cb5e99023844f3ffce127d15e;received=192.168.1.76
From: <sip:525@24.123.23.170>;tag=0e928b1b018f140ea648cfb9cd199547
To: <sip:525@24.123.23.170>;tag=as4477b37f
Call-ID: 55ae66476793@24.123.23.170
CSeq: 39000 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 60
Contact: <sip:525@192.168.1.76:5060>;expires=60
Date: Sat, 10 Mar 2007 17:02:29 GMT
Content-Length: 0


---

[Mar 10 12:02:29] Scheduling destruction of call '55ae66476793@24.123.23.170' in 15000 ms
[Mar 10 12:02:29] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 525
[Mar 10 12:02:29] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/525 - state 1 (Not in use)
[Mar 10 12:02:29] DEBUG[15217]: app_queue.c:500 changethread: Device 'SIP/525' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:02:30] DEBUG[15149]: chan_sip.c:1334 __sip_autodestruct: Auto destroying call '1319586744@192.168.1.97'

[Mar 10 12:02:30] Destroying call '1319586744@192.168.1.97'
[Mar 10 12:02:30] DEBUG[15149]: chan_sip.c:1195 retrans_pkt: SIP TIMER: Rescheduling retransmission #133 (5) SIP/2.0 - 1
[Mar 10 12:02:30] DEBUG[15149]: chan_sip.c:1209 retrans_pkt: ** SIP timers: Rescheduling retransmission 6 to 4000 ms (t1 500 ms (Retrans id #133)) 

[Mar 10 12:02:30] Retransmitting #5 (no NAT) to 192.168.1.85:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKaf8142956f618711e28805bf4c5289dfc;received=192.168.1.85
From: 523 <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f
To: <sip:593@24.123.23.170>;tag=as0da2dd59
Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170
CSeq: 388769 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:593@24.123.23.170>
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 15131 15134 IN IP4 192.168.1.83
s=session
c=IN IP4 192.168.1.83
t=0 0
m=audio 51000 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
[Mar 10 12:02:31] NOTICE[15149]: chan_sip.c:5399 sip_reregister:    -- Re-registration for  3173241052@sip.broadvoice.com
[Mar 10 12:02:31] DEBUG[15149]: chan_sip.c:3168 sip_alloc: Allocating new SIP dialog for 71f439ca643c96aa63da468e6d077ab4@24.123.23.170 - REGISTER (No RTP)
[Mar 10 12:02:31] DEBUG[15149]: chan_sip.c:5564 transmit_register: Scheduled a registration timeout for sip.broadvoice.com id  #156 
[Mar 10 12:02:31] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: REGISTER sip:sip.broadvoice.com SIP/2.0 (39)
[Mar 10 12:02:31] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK056151e5;rport (64)
[Mar 10 12:02:31] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: From: <sip:3173241052@sip.broadvoice.com>;tag=as6174f80c (56)
[Mar 10 12:02:31] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: To: <sip:3173241052@sip.broadvoice.com> (39)
[Mar 10 12:02:31] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Call-ID: 71f439ca643c96aa63da468e6d077ab4@24.123.23.170 (55)
[Mar 10 12:02:31] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: CSeq: 105 REGISTER (18)
[Mar 10 12:02:31] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: User-Agent: Asterisk PBX (24)
[Mar 10 12:02:31] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Max-Forwards: 70 (16)
[Mar 10 12:02:31] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Expires: 120 (12)
[Mar 10 12:02:31] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Contact: <sip:3173241052@24.123.23.170> (39)
[Mar 10 12:02:31] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Event: registration (19)
[Mar 10 12:02:31] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11: Content-Length: 0 (17)
[Mar 10 12:02:31] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 12:  (0)

[Mar 10 12:02:31] REGISTER 12 headers, 0 lines

[Mar 10 12:02:31] REGISTER attempt 1 to 3173241052@sip.broadvoice.com

[Mar 10 12:02:31] Reliably Transmitting (no NAT) to 147.135.12.128:5060:
REGISTER sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK056151e5;rport
From: <sip:3173241052@sip.broadvoice.com>;tag=as6174f80c
To: <sip:3173241052@sip.broadvoice.com>
Call-ID: 71f439ca643c96aa63da468e6d077ab4@24.123.23.170
CSeq: 105 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: <sip:3173241052@24.123.23.170>
Event: registration
Content-Length: 0


---
[Mar 10 12:02:31] DEBUG[15149]: chan_sip.c:1304 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id  #157

[Mar 10 12:02:31] 
<-- SIP read from 147.135.12.128:5060: 
SIP/2.0 200 OK
Call-ID: 71f439ca643c96aa63da468e6d077ab4@24.123.23.170
CSeq: 105 REGISTER
From: <sip:3173241052@sip.broadvoice.com>;tag=as6174f80c
To: <sip:3173241052@sip.broadvoice.com>
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK056151e5
Contact: <sip:3173241052@24.123.23.170>
Expires: 30
Event: registration
Content-Length:    0


[Mar 10 12:02:31] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: SIP/2.0 200 OK (14)
[Mar 10 12:02:31] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Call-ID: 71f439ca643c96aa63da468e6d077ab4@24.123.23.170 (55)
[Mar 10 12:02:31] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: CSeq: 105 REGISTER (18)
[Mar 10 12:02:31] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: From: <sip:3173241052@sip.broadvoice.com>;tag=as6174f80c (56)
[Mar 10 12:02:31] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: To: <sip:3173241052@sip.broadvoice.com> (39)
[Mar 10 12:02:31] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK056151e5 (58)
[Mar 10 12:02:31] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: Contact: <sip:3173241052@24.123.23.170> (39)
[Mar 10 12:02:31] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Expires: 30 (11)
[Mar 10 12:02:31] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Event: registration (19)
[Mar 10 12:02:31] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Content-Length:    0 (20)
[Mar 10 12:02:31] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10:  (0)

[Mar 10 12:02:31] --- (10 headers 0 lines) ---
[Mar 10 12:02:31] DEBUG[15149]: chan_sip.c:1401 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #157
[Mar 10 12:02:31] DEBUG[15149]: chan_sip.c:1412 __sip_ack: Stopping retransmission on '71f439ca643c96aa63da468e6d077ab4@24.123.23.170' of Request 105: Match Found
[Mar 10 12:02:31] DEBUG[15149]: chan_sip.c:9843 handle_response_register: Registration successful
[Mar 10 12:02:31] DEBUG[15149]: chan_sip.c:9845 handle_response_register: Cancelling timeout 156

[Mar 10 12:02:31] Scheduling destruction of call '71f439ca643c96aa63da468e6d077ab4@24.123.23.170' in 32000 ms
[Mar 10 12:02:31] NOTICE[15149]: chan_sip.c:9895 handle_response_register: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 23 s)

[Mar 10 12:02:31] 
<-- SIP read from 192.168.1.86:5060: 


[Mar 10 12:02:31] --- (0 headers 0 lines) Nat keepalive ---

[Mar 10 12:02:33] 
<-- SIP read from 192.168.1.61:5060: 
REGISTER sip:24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bKbf152b2bdda0276b8aab8100b7169a28b
Call-ID: 55ae6647678c@24.123.23.170
CSeq: 61334 REGISTER
From: <sip:524@24.123.23.170>;tag=73601804d4e05da543cbef50f74ca510
To: <sip:524@24.123.23.170>
Contact: <sip:524@192.168.1.61:5060>;expires=60;q=0.50
User-Agent: Uniden SIP Phone p2 Ver BS4.77
Max-Forwards: 70
Content-Length: 0
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK
Authorization: Digest realm="asterisk",   nonce="63f09c99",   algorithm=MD5,   uri="sip:24.123.23.170",   username="524",   response="4ce47e1d1080254df66f29770d96e5d1"


[Mar 10 12:02:33] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34)
[Mar 10 12:02:33] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bKbf152b2bdda0276b8aab8100b7169a28b (82)
[Mar 10 12:02:33] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: Call-ID: 55ae6647678c@24.123.23.170 (35)
[Mar 10 12:02:33] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: CSeq: 61334 REGISTER (20)
[Mar 10 12:02:33] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: From: <sip:524@24.123.23.170>;tag=73601804d4e05da543cbef50f74ca510 (66)
[Mar 10 12:02:33] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: To: <sip:524@24.123.23.170> (27)
[Mar 10 12:02:33] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: Contact: <sip:524@192.168.1.61:5060>;expires=60;q=0.50 (54)
[Mar 10 12:02:33] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42)
[Mar 10 12:02:33] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Max-Forwards: 70 (16)
[Mar 10 12:02:33] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Content-Length: 0 (17)
[Mar 10 12:02:33] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45)
[Mar 10 12:02:33] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62)
[Mar 10 12:02:33] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 12: Authorization: Digest realm="asterisk",   nonce="63f09c99",   algorithm=MD5,   uri="sip:24.123.23.170",   username="524",   response="4ce47e1d1080254df66f29770d96e5d1" (167)
[Mar 10 12:02:33] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 13:  (0)

[Mar 10 12:02:33] --- (13 headers 0 lines) ---
[Mar 10 12:02:33] DEBUG[15149]: chan_sip.c:3168 sip_alloc: Allocating new SIP dialog for 55ae6647678c@24.123.23.170 - REGISTER (No RTP)
[Mar 10 12:02:33] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:02:33] Using latest REGISTER request as basis request

[Mar 10 12:02:33] Sending to 192.168.1.61 : 5060 (non-NAT)

[Mar 10 12:02:33] Transmitting (no NAT) to 192.168.1.61:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bKbf152b2bdda0276b8aab8100b7169a28b;received=192.168.1.61
From: <sip:524@24.123.23.170>;tag=73601804d4e05da543cbef50f74ca510
To: <sip:524@24.123.23.170>
Call-ID: 55ae6647678c@24.123.23.170
CSeq: 61334 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:524@24.123.23.170>
Content-Length: 0


---

[Mar 10 12:02:33] Transmitting (no NAT) to 192.168.1.61:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bKbf152b2bdda0276b8aab8100b7169a28b;received=192.168.1.61
From: <sip:524@24.123.23.170>;tag=73601804d4e05da543cbef50f74ca510
To: <sip:524@24.123.23.170>;tag=as059568ac
Call-ID: 55ae6647678c@24.123.23.170
CSeq: 61334 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="45a8fb68"
Content-Length: 0


---

[Mar 10 12:02:33] Scheduling destruction of call '55ae6647678c@24.123.23.170' in 15000 ms

[Mar 10 12:02:33] 
<-- SIP read from 192.168.1.61:5060: 
REGISTER sip:24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bKk6ed93221a1f2000660a159f17ecb1d0c
CSeq: 61335 REGISTER
Call-ID: 55ae6647678c@24.123.23.170
From: <sip:524@24.123.23.170>;tag=73601804d4e05da543cbef50f74ca510
To: <sip:524@24.123.23.170>
Contact: <sip:524@192.168.1.61:5060>;expires=60;q=0.50
User-Agent: Uniden SIP Phone p2 Ver BS4.77
Max-Forwards: 70
Content-Length: 0
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK
Authorization: Digest realm="asterisk",   nonce="45a8fb68",   algorithm=MD5,   uri="sip:24.123.23.170",   username="524",   response="8134fa0f6abf4654516151bf0d0d7e04"


[Mar 10 12:02:33] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34)
[Mar 10 12:02:33] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bKk6ed93221a1f2000660a159f17ecb1d0c (82)
[Mar 10 12:02:33] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: CSeq: 61335 REGISTER (20)
[Mar 10 12:02:33] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: Call-ID: 55ae6647678c@24.123.23.170 (35)
[Mar 10 12:02:33] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: From: <sip:524@24.123.23.170>;tag=73601804d4e05da543cbef50f74ca510 (66)
[Mar 10 12:02:33] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: To: <sip:524@24.123.23.170> (27)
[Mar 10 12:02:33] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: Contact: <sip:524@192.168.1.61:5060>;expires=60;q=0.50 (54)
[Mar 10 12:02:33] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42)
[Mar 10 12:02:33] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Max-Forwards: 70 (16)
[Mar 10 12:02:33] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Content-Length: 0 (17)
[Mar 10 12:02:33] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45)
[Mar 10 12:02:33] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62)
[Mar 10 12:02:33] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 12: Authorization: Digest realm="asterisk",   nonce="45a8fb68",   algorithm=MD5,   uri="sip:24.123.23.170",   username="524",   response="8134fa0f6abf4654516151bf0d0d7e04" (167)
[Mar 10 12:02:33] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 13:  (0)

[Mar 10 12:02:33] --- (13 headers 0 lines) ---
[Mar 10 12:02:33] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:02:33] Using latest REGISTER request as basis request

[Mar 10 12:02:33] Sending to 192.168.1.61 : 5060 (non-NAT)

[Mar 10 12:02:33] Transmitting (no NAT) to 192.168.1.61:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bKk6ed93221a1f2000660a159f17ecb1d0c;received=192.168.1.61
From: <sip:524@24.123.23.170>;tag=73601804d4e05da543cbef50f74ca510
To: <sip:524@24.123.23.170>
Call-ID: 55ae6647678c@24.123.23.170
CSeq: 61335 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:524@24.123.23.170>
Content-Length: 0


---

[Mar 10 12:02:33] Transmitting (no NAT) to 192.168.1.61:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bKk6ed93221a1f2000660a159f17ecb1d0c;received=192.168.1.61
From: <sip:524@24.123.23.170>;tag=73601804d4e05da543cbef50f74ca510
To: <sip:524@24.123.23.170>;tag=as059568ac
Call-ID: 55ae6647678c@24.123.23.170
CSeq: 61335 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 60
Contact: <sip:524@192.168.1.61:5060>;expires=60
Date: Sat, 10 Mar 2007 17:02:33 GMT
Content-Length: 0


---

[Mar 10 12:02:33] Scheduling destruction of call '55ae6647678c@24.123.23.170' in 15000 ms
[Mar 10 12:02:33] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 524
[Mar 10 12:02:33] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/524 - state 1 (Not in use)
[Mar 10 12:02:33] DEBUG[15218]: app_queue.c:500 changethread: Device 'SIP/524' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:02:33] DEBUG[15149]: chan_sip.c:1334 __sip_autodestruct: Auto destroying call '55ae66469f7f@24.123.23.170'

[Mar 10 12:02:33] Destroying call '55ae66469f7f@24.123.23.170'
[Mar 10 12:02:34] DEBUG[15149]: chan_sip.c:1334 __sip_autodestruct: Auto destroying call '55ae664696c0@24.123.23.170'

[Mar 10 12:02:34] Destroying call '55ae664696c0@24.123.23.170'
[Mar 10 12:02:34] DEBUG[15149]: chan_sip.c:1195 retrans_pkt: SIP TIMER: Rescheduling retransmission #133 (6) SIP/2.0 - 1
[Mar 10 12:02:34] DEBUG[15149]: chan_sip.c:1209 retrans_pkt: ** SIP timers: Rescheduling retransmission 7 to 4000 ms (t1 500 ms (Retrans id #133)) 

[Mar 10 12:02:34] Retransmitting #6 (no NAT) to 192.168.1.85:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKaf8142956f618711e28805bf4c5289dfc;received=192.168.1.85
From: 523 <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f
To: <sip:593@24.123.23.170>;tag=as0da2dd59
Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170
CSeq: 388769 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:593@24.123.23.170>
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 15131 15134 IN IP4 192.168.1.83
s=session
c=IN IP4 192.168.1.83
t=0 0
m=audio 51000 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
[Mar 10 12:02:35] DEBUG[15149]: chan_sip.c:1334 __sip_autodestruct: Auto destroying call '000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165'

[Mar 10 12:02:35] Destroying call '000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165'

[Mar 10 12:02:35] 
<-- SIP read from 192.168.1.95:5060: 
REGISTER sip:24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.95:5060;branch=z9hG4bKy72328fe1513b923f65a8b0f4ff08b808
Call-ID: 55ae66469f11@24.123.23.170
CSeq: 49833 REGISTER
From: <sip:522@24.123.23.170>;tag=cc96e59c46dd8a5e2304d51d822d1a6e
To: <sip:522@24.123.23.170>
Contact: <sip:522@192.168.1.95:5060>;expires=60;q=0.50
User-Agent: Uniden SIP Phone p2 Ver BS4.77
Max-Forwards: 70
Content-Length: 0
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK
Authorization: Digest realm="asterisk",   nonce="034d5f34",   algorithm=MD5,   uri="sip:24.123.23.170",   username="522",   response="6a8b92852f26228fdecd6798da79287c"


[Mar 10 12:02:35] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34)
[Mar 10 12:02:35] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.95:5060;branch=z9hG4bKy72328fe1513b923f65a8b0f4ff08b808 (82)
[Mar 10 12:02:35] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: Call-ID: 55ae66469f11@24.123.23.170 (35)
[Mar 10 12:02:35] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: CSeq: 49833 REGISTER (20)
[Mar 10 12:02:35] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: From: <sip:522@24.123.23.170>;tag=cc96e59c46dd8a5e2304d51d822d1a6e (66)
[Mar 10 12:02:35] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: To: <sip:522@24.123.23.170> (27)
[Mar 10 12:02:35] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: Contact: <sip:522@192.168.1.95:5060>;expires=60;q=0.50 (54)
[Mar 10 12:02:35] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42)
[Mar 10 12:02:35] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Max-Forwards: 70 (16)
[Mar 10 12:02:35] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Content-Length: 0 (17)
[Mar 10 12:02:35] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45)
[Mar 10 12:02:35] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62)
[Mar 10 12:02:35] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 12: Authorization: Digest realm="asterisk",   nonce="034d5f34",   algorithm=MD5,   uri="sip:24.123.23.170",   username="522",   response="6a8b92852f26228fdecd6798da79287c" (167)
[Mar 10 12:02:35] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 13:  (0)

[Mar 10 12:02:35] --- (13 headers 0 lines) ---
[Mar 10 12:02:35] DEBUG[15149]: chan_sip.c:3168 sip_alloc: Allocating new SIP dialog for 55ae66469f11@24.123.23.170 - REGISTER (No RTP)
[Mar 10 12:02:35] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:02:35] Using latest REGISTER request as basis request

[Mar 10 12:02:35] Sending to 192.168.1.95 : 5060 (non-NAT)

[Mar 10 12:02:35] Transmitting (no NAT) to 192.168.1.95:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.95:5060;branch=z9hG4bKy72328fe1513b923f65a8b0f4ff08b808;received=192.168.1.95
From: <sip:522@24.123.23.170>;tag=cc96e59c46dd8a5e2304d51d822d1a6e
To: <sip:522@24.123.23.170>
Call-ID: 55ae66469f11@24.123.23.170
CSeq: 49833 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:522@24.123.23.170>
Content-Length: 0


---

[Mar 10 12:02:35] Transmitting (no NAT) to 192.168.1.95:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.95:5060;branch=z9hG4bKy72328fe1513b923f65a8b0f4ff08b808;received=192.168.1.95
From: <sip:522@24.123.23.170>;tag=cc96e59c46dd8a5e2304d51d822d1a6e
To: <sip:522@24.123.23.170>;tag=as4ae074dd
Call-ID: 55ae66469f11@24.123.23.170
CSeq: 49833 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1a40fe0d"
Content-Length: 0


---

[Mar 10 12:02:35] Scheduling destruction of call '55ae66469f11@24.123.23.170' in 15000 ms

[Mar 10 12:02:35] 
<-- SIP read from 192.168.1.95:5060: 
REGISTER sip:24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.95:5060;branch=z9hG4bKhcb495cd03d681fab0ede94c3299204b3
CSeq: 49834 REGISTER
Call-ID: 55ae66469f11@24.123.23.170
From: <sip:522@24.123.23.170>;tag=cc96e59c46dd8a5e2304d51d822d1a6e
To: <sip:522@24.123.23.170>
Contact: <sip:522@192.168.1.95:5060>;expires=60;q=0.50
User-Agent: Uniden SIP Phone p2 Ver BS4.77
Max-Forwards: 70
Content-Length: 0
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK
Authorization: Digest realm="asterisk",   nonce="1a40fe0d",   algorithm=MD5,   uri="sip:24.123.23.170",   username="522",   response="3c5ff6dfa5bc1c9eaa5cdda6429a920c"


[Mar 10 12:02:35] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34)
[Mar 10 12:02:35] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.95:5060;branch=z9hG4bKhcb495cd03d681fab0ede94c3299204b3 (82)
[Mar 10 12:02:35] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: CSeq: 49834 REGISTER (20)
[Mar 10 12:02:35] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: Call-ID: 55ae66469f11@24.123.23.170 (35)
[Mar 10 12:02:35] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: From: <sip:522@24.123.23.170>;tag=cc96e59c46dd8a5e2304d51d822d1a6e (66)
[Mar 10 12:02:35] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: To: <sip:522@24.123.23.170> (27)
[Mar 10 12:02:35] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: Contact: <sip:522@192.168.1.95:5060>;expires=60;q=0.50 (54)
[Mar 10 12:02:35] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42)
[Mar 10 12:02:35] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Max-Forwards: 70 (16)
[Mar 10 12:02:35] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Content-Length: 0 (17)
[Mar 10 12:02:35] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45)
[Mar 10 12:02:35] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62)
[Mar 10 12:02:35] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 12: Authorization: Digest realm="asterisk",   nonce="1a40fe0d",   algorithm=MD5,   uri="sip:24.123.23.170",   username="522",   response="3c5ff6dfa5bc1c9eaa5cdda6429a920c" (167)
[Mar 10 12:02:35] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 13:  (0)

[Mar 10 12:02:35] --- (13 headers 0 lines) ---
[Mar 10 12:02:35] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:02:35] Using latest REGISTER request as basis request

[Mar 10 12:02:35] Sending to 192.168.1.95 : 5060 (non-NAT)

[Mar 10 12:02:35] Transmitting (no NAT) to 192.168.1.95:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.95:5060;branch=z9hG4bKhcb495cd03d681fab0ede94c3299204b3;received=192.168.1.95
From: <sip:522@24.123.23.170>;tag=cc96e59c46dd8a5e2304d51d822d1a6e
To: <sip:522@24.123.23.170>
Call-ID: 55ae66469f11@24.123.23.170
CSeq: 49834 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:522@24.123.23.170>
Content-Length: 0


---

[Mar 10 12:02:35] Transmitting (no NAT) to 192.168.1.95:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.95:5060;branch=z9hG4bKhcb495cd03d681fab0ede94c3299204b3;received=192.168.1.95
From: <sip:522@24.123.23.170>;tag=cc96e59c46dd8a5e2304d51d822d1a6e
To: <sip:522@24.123.23.170>;tag=as4ae074dd
Call-ID: 55ae66469f11@24.123.23.170
CSeq: 49834 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 60
Contact: <sip:522@192.168.1.95:5060>;expires=60
Date: Sat, 10 Mar 2007 17:02:35 GMT
Content-Length: 0


---

[Mar 10 12:02:35] Scheduling destruction of call '55ae66469f11@24.123.23.170' in 15000 ms
[Mar 10 12:02:35] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 522
[Mar 10 12:02:35] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/522 - state 1 (Not in use)
[Mar 10 12:02:35] DEBUG[15219]: app_queue.c:500 changethread: Device 'SIP/522' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.

[Mar 10 12:02:36] 
<-- SIP read from 192.168.1.66:5060: 
REGISTER sip:24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bKh27371b83f38c534128aa82ac841a7c44
Call-ID: 55ae6647666e@24.123.23.170
CSeq: 24025 REGISTER
From: <sip:526@24.123.23.170>;tag=d057287ee0d1cea51492a78e4e28fb98
To: <sip:526@24.123.23.170>
Contact: <sip:526@192.168.1.66:5060>;expires=60;q=0.50
User-Agent: Uniden SIP Phone p2 Ver BS4.77
Max-Forwards: 70
Content-Length: 0
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK
Authorization: Digest realm="asterisk",   nonce="0e84da27",   algorithm=MD5,   uri="sip:24.123.23.170",   username="526",   response="0fd939892aea60351a5416f9b923714a"


[Mar 10 12:02:36] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34)
[Mar 10 12:02:36] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bKh27371b83f38c534128aa82ac841a7c44 (82)
[Mar 10 12:02:36] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: Call-ID: 55ae6647666e@24.123.23.170 (35)
[Mar 10 12:02:36] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: CSeq: 24025 REGISTER (20)
[Mar 10 12:02:36] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: From: <sip:526@24.123.23.170>;tag=d057287ee0d1cea51492a78e4e28fb98 (66)
[Mar 10 12:02:36] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: To: <sip:526@24.123.23.170> (27)
[Mar 10 12:02:36] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: Contact: <sip:526@192.168.1.66:5060>;expires=60;q=0.50 (54)
[Mar 10 12:02:36] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42)
[Mar 10 12:02:36] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Max-Forwards: 70 (16)
[Mar 10 12:02:36] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Content-Length: 0 (17)
[Mar 10 12:02:36] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45)
[Mar 10 12:02:36] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62)
[Mar 10 12:02:36] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 12: Authorization: Digest realm="asterisk",   nonce="0e84da27",   algorithm=MD5,   uri="sip:24.123.23.170",   username="526",   response="0fd939892aea60351a5416f9b923714a" (167)
[Mar 10 12:02:36] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 13:  (0)

[Mar 10 12:02:36] --- (13 headers 0 lines) ---
[Mar 10 12:02:36] DEBUG[15149]: chan_sip.c:3168 sip_alloc: Allocating new SIP dialog for 55ae6647666e@24.123.23.170 - REGISTER (No RTP)
[Mar 10 12:02:36] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:02:36] Using latest REGISTER request as basis request

[Mar 10 12:02:36] Sending to 192.168.1.66 : 5060 (non-NAT)

[Mar 10 12:02:36] Transmitting (no NAT) to 192.168.1.66:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bKh27371b83f38c534128aa82ac841a7c44;received=192.168.1.66
From: <sip:526@24.123.23.170>;tag=d057287ee0d1cea51492a78e4e28fb98
To: <sip:526@24.123.23.170>
Call-ID: 55ae6647666e@24.123.23.170
CSeq: 24025 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:526@24.123.23.170>
Content-Length: 0


---

[Mar 10 12:02:36] Transmitting (no NAT) to 192.168.1.66:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bKh27371b83f38c534128aa82ac841a7c44;received=192.168.1.66
From: <sip:526@24.123.23.170>;tag=d057287ee0d1cea51492a78e4e28fb98
To: <sip:526@24.123.23.170>;tag=as1ca6cd5c
Call-ID: 55ae6647666e@24.123.23.170
CSeq: 24025 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0788d769"
Content-Length: 0


---

[Mar 10 12:02:36] Scheduling destruction of call '55ae6647666e@24.123.23.170' in 15000 ms

[Mar 10 12:02:36] 
<-- SIP read from 192.168.1.66:5060: 
REGISTER sip:24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bKlf9b215afdca8a508dd4148e2446edd39
CSeq: 24026 REGISTER
Call-ID: 55ae6647666e@24.123.23.170
From: <sip:526@24.123.23.170>;tag=d057287ee0d1cea51492a78e4e28fb98
To: <sip:526@24.123.23.170>
Contact: <sip:526@192.168.1.66:5060>;expires=60;q=0.50
User-Agent: Uniden SIP Phone p2 Ver BS4.77
Max-Forwards: 70
Content-Length: 0
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK
Authorization: Digest realm="asterisk",   nonce="0788d769",   algorithm=MD5,   uri="sip:24.123.23.170",   username="526",   response="0d5625bd6c3d591cebf55e11d862e631"


[Mar 10 12:02:36] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34)
[Mar 10 12:02:36] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bKlf9b215afdca8a508dd4148e2446edd39 (82)
[Mar 10 12:02:36] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: CSeq: 24026 REGISTER (20)
[Mar 10 12:02:36] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: Call-ID: 55ae6647666e@24.123.23.170 (35)
[Mar 10 12:02:36] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: From: <sip:526@24.123.23.170>;tag=d057287ee0d1cea51492a78e4e28fb98 (66)
[Mar 10 12:02:36] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: To: <sip:526@24.123.23.170> (27)
[Mar 10 12:02:36] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: Contact: <sip:526@192.168.1.66:5060>;expires=60;q=0.50 (54)
[Mar 10 12:02:36] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42)
[Mar 10 12:02:36] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Max-Forwards: 70 (16)
[Mar 10 12:02:36] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Content-Length: 0 (17)
[Mar 10 12:02:36] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45)
[Mar 10 12:02:36] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62)
[Mar 10 12:02:36] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 12: Authorization: Digest realm="asterisk",   nonce="0788d769",   algorithm=MD5,   uri="sip:24.123.23.170",   username="526",   response="0d5625bd6c3d591cebf55e11d862e631" (167)
[Mar 10 12:02:36] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 13:  (0)

[Mar 10 12:02:36] --- (13 headers 0 lines) ---
[Mar 10 12:02:36] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:02:36] Using latest REGISTER request as basis request

[Mar 10 12:02:36] Sending to 192.168.1.66 : 5060 (non-NAT)

[Mar 10 12:02:36] Transmitting (no NAT) to 192.168.1.66:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bKlf9b215afdca8a508dd4148e2446edd39;received=192.168.1.66
From: <sip:526@24.123.23.170>;tag=d057287ee0d1cea51492a78e4e28fb98
To: <sip:526@24.123.23.170>
Call-ID: 55ae6647666e@24.123.23.170
CSeq: 24026 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:526@24.123.23.170>
Content-Length: 0


---

[Mar 10 12:02:36]     -- Registered SIP '526' at 192.168.1.66 port 5060 expires 60

[Mar 10 12:02:36] Transmitting (no NAT) to 192.168.1.66:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bKlf9b215afdca8a508dd4148e2446edd39;received=192.168.1.66
From: <sip:526@24.123.23.170>;tag=d057287ee0d1cea51492a78e4e28fb98
To: <sip:526@24.123.23.170>;tag=as1ca6cd5c
Call-ID: 55ae6647666e@24.123.23.170
CSeq: 24026 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 60
Contact: <sip:526@192.168.1.66:5060>;expires=60
Date: Sat, 10 Mar 2007 17:02:36 GMT
Content-Length: 0


---

[Mar 10 12:02:36] Scheduling destruction of call '55ae6647666e@24.123.23.170' in 15000 ms
[Mar 10 12:02:36] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 526
[Mar 10 12:02:36] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/526 - state 1 (Not in use)
[Mar 10 12:02:36] DEBUG[15220]: app_queue.c:500 changethread: Device 'SIP/526' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.

[Mar 10 12:02:37] 
<-- SIP read from 192.168.1.165:50208: 
REGISTER sip:24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK04876099
From: <sip:514@24.123.23.170;user=phone>
To: <sip:514@24.123.23.170;user=phone>
Call-ID: 000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165
CSeq: 11568 REGISTER
User-Agent: CSCO/7
Contact: <sip:514@192.168.1.165:5060>
Content-Length: 0
Expires: 60


[Mar 10 12:02:37] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34)
[Mar 10 12:02:37] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK04876099 (58)
[Mar 10 12:02:37] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: From: <sip:514@24.123.23.170;user=phone> (40)
[Mar 10 12:02:37] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: To: <sip:514@24.123.23.170;user=phone> (38)
[Mar 10 12:02:37] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Call-ID: 000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165 (58)
[Mar 10 12:02:37] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: CSeq: 11568 REGISTER (20)
[Mar 10 12:02:37] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: User-Agent: CSCO/7 (18)
[Mar 10 12:02:37] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Contact: <sip:514@192.168.1.165:5060> (37)
[Mar 10 12:02:37] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Content-Length: 0 (17)
[Mar 10 12:02:37] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Expires: 60 (11)
[Mar 10 12:02:37] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10:  (0)

[Mar 10 12:02:37] --- (10 headers 0 lines) ---
[Mar 10 12:02:37] DEBUG[15149]: chan_sip.c:3168 sip_alloc: Allocating new SIP dialog for 000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165 - REGISTER (No RTP)
[Mar 10 12:02:37] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:02:37] Using latest REGISTER request as basis request

[Mar 10 12:02:37] Sending to 192.168.1.165 : 5060 (non-NAT)

[Mar 10 12:02:37] Transmitting (no NAT) to 192.168.1.165:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK04876099;received=192.168.1.165
From: <sip:514@24.123.23.170;user=phone>
To: <sip:514@24.123.23.170;user=phone>
Call-ID: 000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165
CSeq: 11568 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:514@24.123.23.170>
Content-Length: 0


---

[Mar 10 12:02:37] Transmitting (no NAT) to 192.168.1.165:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK04876099;received=192.168.1.165
From: <sip:514@24.123.23.170;user=phone>
To: <sip:514@24.123.23.170;user=phone>;tag=as6341af51
Call-ID: 000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165
CSeq: 11568 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="74af6685"
Content-Length: 0


---

[Mar 10 12:02:37] Scheduling destruction of call '000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165' in 15000 ms

[Mar 10 12:02:37] 
<-- SIP read from 192.168.1.165:50208: 
REGISTER sip:24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK20ed8192
From: <sip:514@24.123.23.170;user=phone>
To: <sip:514@24.123.23.170;user=phone>
Call-ID: 000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165
CSeq: 11569 REGISTER
User-Agent: CSCO/7
Contact: <sip:514@192.168.1.165:5060>
Authorization: Digest username="514",realm="asterisk",uri="sip:24.123.23.170",response="541e2570581a8dc0ba48a41433da6bdd",nonce="74af6685",algorithm=MD5
Content-Length: 0
Expires: 60


[Mar 10 12:02:37] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34)
[Mar 10 12:02:37] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK20ed8192 (58)
[Mar 10 12:02:37] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: From: <sip:514@24.123.23.170;user=phone> (40)
[Mar 10 12:02:37] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: To: <sip:514@24.123.23.170;user=phone> (38)
[Mar 10 12:02:37] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Call-ID: 000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165 (58)
[Mar 10 12:02:37] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: CSeq: 11569 REGISTER (20)
[Mar 10 12:02:37] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: User-Agent: CSCO/7 (18)
[Mar 10 12:02:37] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Contact: <sip:514@192.168.1.165:5060> (37)
[Mar 10 12:02:37] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Authorization: Digest username="514",realm="asterisk",uri="sip:24.123.23.170",response="541e2570581a8dc0ba48a41433da6bdd",nonce="74af6685",algorithm=MD5 (152)
[Mar 10 12:02:37] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Content-Length: 0 (17)
[Mar 10 12:02:37] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Expires: 60 (11)
[Mar 10 12:02:37] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11:  (0)

[Mar 10 12:02:37] --- (11 headers 0 lines) ---
[Mar 10 12:02:37] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:02:37] Using latest REGISTER request as basis request

[Mar 10 12:02:37] Sending to 192.168.1.165 : 5060 (non-NAT)

[Mar 10 12:02:37] Transmitting (no NAT) to 192.168.1.165:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK20ed8192;received=192.168.1.165
From: <sip:514@24.123.23.170;user=phone>
To: <sip:514@24.123.23.170;user=phone>
Call-ID: 000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165
CSeq: 11569 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:514@24.123.23.170>
Content-Length: 0


---

[Mar 10 12:02:37] Transmitting (no NAT) to 192.168.1.165:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK20ed8192;received=192.168.1.165
From: <sip:514@24.123.23.170;user=phone>
To: <sip:514@24.123.23.170;user=phone>;tag=as6341af51
Call-ID: 000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165
CSeq: 11569 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 60
Contact: <sip:514@192.168.1.165:5060>;expires=60
Date: Sat, 10 Mar 2007 17:02:37 GMT
Content-Length: 0


---

[Mar 10 12:02:37] Scheduling destruction of call '000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165' in 15000 ms
[Mar 10 12:02:37] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 514
[Mar 10 12:02:37] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/514 - state 1 (Not in use)
[Mar 10 12:02:37] DEBUG[15221]: app_queue.c:500 changethread: Device 'SIP/514' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:02:38] DEBUG[15149]: chan_sip.c:1334 __sip_autodestruct: Auto destroying call '000ff78d-ebb20002-23a7f663-5bd85286@192.168.1.165'

[Mar 10 12:02:38] Destroying call '000ff78d-ebb20002-23a7f663-5bd85286@192.168.1.165'
[Mar 10 12:02:38] WARNING[15149]: chan_sip.c:1228 retrans_pkt: Maximum retries exceeded on transmission 5d28f46f54fd450475a9f02419f99674@24.123.23.170 for seqno 388769 (Non-critical Response)

[Mar 10 12:02:38] 
<-- SIP read from 192.168.1.165:50207: 
REGISTER sip:24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK3a3fe926
From: <sip:513@24.123.23.170;user=phone>
To: <sip:513@24.123.23.170;user=phone>
Call-ID: 000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165
CSeq: 11567 REGISTER
User-Agent: CSCO/7
Contact: <sip:513@192.168.1.165:5060>
Content-Length: 0
Expires: 60


[Mar 10 12:02:38] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34)
[Mar 10 12:02:38] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK3a3fe926 (58)
[Mar 10 12:02:38] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: From: <sip:513@24.123.23.170;user=phone> (40)
[Mar 10 12:02:38] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: To: <sip:513@24.123.23.170;user=phone> (38)
[Mar 10 12:02:38] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Call-ID: 000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165 (58)
[Mar 10 12:02:38] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: CSeq: 11567 REGISTER (20)
[Mar 10 12:02:38] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: User-Agent: CSCO/7 (18)
[Mar 10 12:02:38] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Contact: <sip:513@192.168.1.165:5060> (37)
[Mar 10 12:02:38] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Content-Length: 0 (17)
[Mar 10 12:02:38] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Expires: 60 (11)
[Mar 10 12:02:38] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10:  (0)

[Mar 10 12:02:38] --- (10 headers 0 lines) ---
[Mar 10 12:02:38] DEBUG[15149]: chan_sip.c:3168 sip_alloc: Allocating new SIP dialog for 000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165 - REGISTER (No RTP)
[Mar 10 12:02:38] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:02:38] Using latest REGISTER request as basis request

[Mar 10 12:02:38] Sending to 192.168.1.165 : 5060 (non-NAT)

[Mar 10 12:02:38] Transmitting (no NAT) to 192.168.1.165:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK3a3fe926;received=192.168.1.165
From: <sip:513@24.123.23.170;user=phone>
To: <sip:513@24.123.23.170;user=phone>
Call-ID: 000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165
CSeq: 11567 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:513@24.123.23.170>
Content-Length: 0


---

[Mar 10 12:02:38] Transmitting (no NAT) to 192.168.1.165:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK3a3fe926;received=192.168.1.165
From: <sip:513@24.123.23.170;user=phone>
To: <sip:513@24.123.23.170;user=phone>;tag=as5c6a2f2b
Call-ID: 000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165
CSeq: 11567 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="09c9b778"
Content-Length: 0


---

[Mar 10 12:02:38] Scheduling destruction of call '000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165' in 15000 ms

[Mar 10 12:02:38] 
<-- SIP read from 192.168.1.97:5060: 


[Mar 10 12:02:38] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0:  (0)

[Mar 10 12:02:38] --- (0 headers 0 lines) Nat keepalive ---

[Mar 10 12:02:38] 
<-- SIP read from 192.168.1.165:50207: 
REGISTER sip:24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK038ca36a
From: <sip:513@24.123.23.170;user=phone>
To: <sip:513@24.123.23.170;user=phone>
Call-ID: 000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165
CSeq: 11568 REGISTER
User-Agent: CSCO/7
Contact: <sip:513@192.168.1.165:5060>
Authorization: Digest username="513",realm="asterisk",uri="sip:24.123.23.170",response="58e98701f48169911162d4f9534bef84",nonce="09c9b778",algorithm=MD5
Content-Length: 0
Expires: 60


[Mar 10 12:02:38] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34)
[Mar 10 12:02:38] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK038ca36a (58)
[Mar 10 12:02:38] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: From: <sip:513@24.123.23.170;user=phone> (40)
[Mar 10 12:02:38] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: To: <sip:513@24.123.23.170;user=phone> (38)
[Mar 10 12:02:38] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Call-ID: 000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165 (58)
[Mar 10 12:02:38] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: CSeq: 11568 REGISTER (20)
[Mar 10 12:02:38] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: User-Agent: CSCO/7 (18)
[Mar 10 12:02:38] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Contact: <sip:513@192.168.1.165:5060> (37)
[Mar 10 12:02:38] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Authorization: Digest username="513",realm="asterisk",uri="sip:24.123.23.170",response="58e98701f48169911162d4f9534bef84",nonce="09c9b778",algorithm=MD5 (152)
[Mar 10 12:02:38] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Content-Length: 0 (17)
[Mar 10 12:02:38] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Expires: 60 (11)
[Mar 10 12:02:38] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11:  (0)

[Mar 10 12:02:38] --- (11 headers 0 lines) ---
[Mar 10 12:02:38] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:02:38] Using latest REGISTER request as basis request

[Mar 10 12:02:38] Sending to 192.168.1.165 : 5060 (non-NAT)

[Mar 10 12:02:38] Transmitting (no NAT) to 192.168.1.165:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK038ca36a;received=192.168.1.165
From: <sip:513@24.123.23.170;user=phone>
To: <sip:513@24.123.23.170;user=phone>
Call-ID: 000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165
CSeq: 11568 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:513@24.123.23.170>
Content-Length: 0


---

[Mar 10 12:02:38] Transmitting (no NAT) to 192.168.1.165:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK038ca36a;received=192.168.1.165
From: <sip:513@24.123.23.170;user=phone>
To: <sip:513@24.123.23.170;user=phone>;tag=as5c6a2f2b
Call-ID: 000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165
CSeq: 11568 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 60
Contact: <sip:513@192.168.1.165:5060>;expires=60
Date: Sat, 10 Mar 2007 17:02:38 GMT
Content-Length: 0


---

[Mar 10 12:02:38] Scheduling destruction of call '000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165' in 15000 ms
[Mar 10 12:02:38] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 513
[Mar 10 12:02:38] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/513 - state 1 (Not in use)
[Mar 10 12:02:38] DEBUG[15222]: app_queue.c:500 changethread: Device 'SIP/513' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:02:39] DEBUG[15149]: chan_sip.c:1334 __sip_autodestruct: Auto destroying call '71f439ca643c96aa63da468e6d077ab4@24.123.23.170'

[Mar 10 12:02:39] Destroying call '71f439ca643c96aa63da468e6d077ab4@24.123.23.170'

[Mar 10 12:02:40] 
<-- SIP read from 192.168.1.85:5060: 
REGISTER sip:24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKmba542c2bd99a6abe2f5c64710a559b2f
Call-ID: 55ae664767e0@24.123.23.170
CSeq: 66109 REGISTER
From: <sip:523@24.123.23.170>;tag=95e4503d50eaffc6006341712dcea52c
To: <sip:523@24.123.23.170>
Contact: <sip:523@192.168.1.85:5060>;expires=60;q=0.50
User-Agent: Uniden SIP Phone p2 Ver BS4.77
Max-Forwards: 70
Content-Length: 0
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK
Authorization: Digest realm="asterisk",   nonce="5b36e87b",   algorithm=MD5,   uri="sip:24.123.23.170",   username="523",   response="fa455ca3792c38ce23c01c9190dc6996"


[Mar 10 12:02:40] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34)
[Mar 10 12:02:40] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKmba542c2bd99a6abe2f5c64710a559b2f (82)
[Mar 10 12:02:40] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: Call-ID: 55ae664767e0@24.123.23.170 (35)
[Mar 10 12:02:40] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: CSeq: 66109 REGISTER (20)
[Mar 10 12:02:40] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: From: <sip:523@24.123.23.170>;tag=95e4503d50eaffc6006341712dcea52c (66)
[Mar 10 12:02:40] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: To: <sip:523@24.123.23.170> (27)
[Mar 10 12:02:40] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: Contact: <sip:523@192.168.1.85:5060>;expires=60;q=0.50 (54)
[Mar 10 12:02:40] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42)
[Mar 10 12:02:40] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Max-Forwards: 70 (16)
[Mar 10 12:02:40] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Content-Length: 0 (17)
[Mar 10 12:02:40] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45)
[Mar 10 12:02:40] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62)
[Mar 10 12:02:40] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 12: Authorization: Digest realm="asterisk",   nonce="5b36e87b",   algorithm=MD5,   uri="sip:24.123.23.170",   username="523",   response="fa455ca3792c38ce23c01c9190dc6996" (167)
[Mar 10 12:02:40] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 13:  (0)

[Mar 10 12:02:40] --- (13 headers 0 lines) ---
[Mar 10 12:02:40] DEBUG[15149]: chan_sip.c:3168 sip_alloc: Allocating new SIP dialog for 55ae664767e0@24.123.23.170 - REGISTER (No RTP)
[Mar 10 12:02:40] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:02:40] Using latest REGISTER request as basis request

[Mar 10 12:02:40] Sending to 192.168.1.85 : 5060 (non-NAT)

[Mar 10 12:02:40] Transmitting (no NAT) to 192.168.1.85:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKmba542c2bd99a6abe2f5c64710a559b2f;received=192.168.1.85
From: <sip:523@24.123.23.170>;tag=95e4503d50eaffc6006341712dcea52c
To: <sip:523@24.123.23.170>
Call-ID: 55ae664767e0@24.123.23.170
CSeq: 66109 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:523@24.123.23.170>
Content-Length: 0


---

[Mar 10 12:02:40] Transmitting (no NAT) to 192.168.1.85:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKmba542c2bd99a6abe2f5c64710a559b2f;received=192.168.1.85
From: <sip:523@24.123.23.170>;tag=95e4503d50eaffc6006341712dcea52c
To: <sip:523@24.123.23.170>;tag=as762fe8bd
Call-ID: 55ae664767e0@24.123.23.170
CSeq: 66109 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="66d7d99b"
Content-Length: 0


---

[Mar 10 12:02:40] Scheduling destruction of call '55ae664767e0@24.123.23.170' in 15000 ms

[Mar 10 12:02:40] 
<-- SIP read from 192.168.1.85:5060: 
REGISTER sip:24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKw1c7c2395b4c88bc510757c0c98522824
CSeq: 66110 REGISTER
Call-ID: 55ae664767e0@24.123.23.170
From: <sip:523@24.123.23.170>;tag=95e4503d50eaffc6006341712dcea52c
To: <sip:523@24.123.23.170>
Contact: <sip:523@192.168.1.85:5060>;expires=60;q=0.50
User-Agent: Uniden SIP Phone p2 Ver BS4.77
Max-Forwards: 70
Content-Length: 0
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK
Authorization: Digest realm="asterisk",   nonce="66d7d99b",   algorithm=MD5,   uri="sip:24.123.23.170",   username="523",   response="832ac5a76256b461d2e5aa8b70f8ac1d"


[Mar 10 12:02:40] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34)
[Mar 10 12:02:40] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKw1c7c2395b4c88bc510757c0c98522824 (82)
[Mar 10 12:02:40] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: CSeq: 66110 REGISTER (20)
[Mar 10 12:02:40] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: Call-ID: 55ae664767e0@24.123.23.170 (35)
[Mar 10 12:02:40] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: From: <sip:523@24.123.23.170>;tag=95e4503d50eaffc6006341712dcea52c (66)
[Mar 10 12:02:40] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: To: <sip:523@24.123.23.170> (27)
[Mar 10 12:02:40] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: Contact: <sip:523@192.168.1.85:5060>;expires=60;q=0.50 (54)
[Mar 10 12:02:40] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42)
[Mar 10 12:02:40] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Max-Forwards: 70 (16)
[Mar 10 12:02:40] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Content-Length: 0 (17)
[Mar 10 12:02:40] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45)
[Mar 10 12:02:40] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62)
[Mar 10 12:02:40] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 12: Authorization: Digest realm="asterisk",   nonce="66d7d99b",   algorithm=MD5,   uri="sip:24.123.23.170",   username="523",   response="832ac5a76256b461d2e5aa8b70f8ac1d" (167)
[Mar 10 12:02:40] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 13:  (0)

[Mar 10 12:02:40] --- (13 headers 0 lines) ---
[Mar 10 12:02:40] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:02:40] Using latest REGISTER request as basis request

[Mar 10 12:02:40] Sending to 192.168.1.85 : 5060 (non-NAT)

[Mar 10 12:02:40] Transmitting (no NAT) to 192.168.1.85:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKw1c7c2395b4c88bc510757c0c98522824;received=192.168.1.85
From: <sip:523@24.123.23.170>;tag=95e4503d50eaffc6006341712dcea52c
To: <sip:523@24.123.23.170>
Call-ID: 55ae664767e0@24.123.23.170
CSeq: 66110 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:523@24.123.23.170>
Content-Length: 0


---

[Mar 10 12:02:40] Transmitting (no NAT) to 192.168.1.85:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKw1c7c2395b4c88bc510757c0c98522824;received=192.168.1.85
From: <sip:523@24.123.23.170>;tag=95e4503d50eaffc6006341712dcea52c
To: <sip:523@24.123.23.170>;tag=as762fe8bd
Call-ID: 55ae664767e0@24.123.23.170
CSeq: 66110 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 60
Contact: <sip:523@192.168.1.85:5060>;expires=60
Date: Sat, 10 Mar 2007 17:02:40 GMT
Content-Length: 0


---

[Mar 10 12:02:40] Scheduling destruction of call '55ae664767e0@24.123.23.170' in 15000 ms
[Mar 10 12:02:40] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 523
[Mar 10 12:02:40] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/523 - state 2 (In use)
[Mar 10 12:02:40] DEBUG[15223]: app_queue.c:500 changethread: Device 'SIP/523' changed to state '2' (In use) but we don't care because they're not a member of any queue.

[Mar 10 12:02:41] 
<-- SIP read from 68.58.36.157:5060: 


[Mar 10 12:02:41] --- (0 headers 0 lines) Nat keepalive ---

[Mar 10 12:02:41] 
<-- SIP read from 192.168.1.165:50206: 
REGISTER sip:24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK38c44cb7
From: <sip:512@24.123.23.170;user=phone>
To: <sip:512@24.123.23.170;user=phone>
Call-ID: 000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165
CSeq: 11588 REGISTER
User-Agent: CSCO/7
Contact: <sip:512@192.168.1.165:5060>
Content-Length: 0
Expires: 60


[Mar 10 12:02:41] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34)
[Mar 10 12:02:41] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK38c44cb7 (58)
[Mar 10 12:02:41] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: From: <sip:512@24.123.23.170;user=phone> (40)
[Mar 10 12:02:41] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: To: <sip:512@24.123.23.170;user=phone> (38)
[Mar 10 12:02:41] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Call-ID: 000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165 (58)
[Mar 10 12:02:41] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: CSeq: 11588 REGISTER (20)
[Mar 10 12:02:41] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: User-Agent: CSCO/7 (18)
[Mar 10 12:02:41] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Contact: <sip:512@192.168.1.165:5060> (37)
[Mar 10 12:02:41] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Content-Length: 0 (17)
[Mar 10 12:02:41] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Expires: 60 (11)
[Mar 10 12:02:41] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10:  (0)

[Mar 10 12:02:41] --- (10 headers 0 lines) ---
[Mar 10 12:02:41] DEBUG[15149]: chan_sip.c:3168 sip_alloc: Allocating new SIP dialog for 000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165 - REGISTER (No RTP)
[Mar 10 12:02:41] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:02:41] Using latest REGISTER request as basis request

[Mar 10 12:02:41] Sending to 192.168.1.165 : 5060 (non-NAT)

[Mar 10 12:02:41] Transmitting (no NAT) to 192.168.1.165:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK38c44cb7;received=192.168.1.165
From: <sip:512@24.123.23.170;user=phone>
To: <sip:512@24.123.23.170;user=phone>
Call-ID: 000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165
CSeq: 11588 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:512@24.123.23.170>
Content-Length: 0


---

[Mar 10 12:02:41] Transmitting (no NAT) to 192.168.1.165:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK38c44cb7;received=192.168.1.165
From: <sip:512@24.123.23.170;user=phone>
To: <sip:512@24.123.23.170;user=phone>;tag=as75433f9c
Call-ID: 000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165
CSeq: 11588 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="30041a4b"
Content-Length: 0


---

[Mar 10 12:02:41] Scheduling destruction of call '000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165' in 15000 ms
[Mar 10 12:02:41] DEBUG[15136]: chan_iax2.c:9448 iax2_devicestate: Checking device state for device 606
[Mar 10 12:02:41] DEBUG[15136]: chan_iax2.c:9456 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 606? addr=-1509840704, defaddr=0 maxms=0, lastms=0
[Mar 10 12:02:41] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for IAX2/606 - state 1 (Not in use)
[Mar 10 12:02:41] DEBUG[15224]: app_queue.c:500 changethread: Device 'IAX2/606' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:02:41] DEBUG[15136]: chan_iax2.c:9448 iax2_devicestate: Checking device state for device 606
[Mar 10 12:02:41] DEBUG[15136]: chan_iax2.c:9456 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 606? addr=-1509840704, defaddr=0 maxms=0, lastms=0
[Mar 10 12:02:41] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for IAX2/606 - state 1 (Not in use)
[Mar 10 12:02:41] DEBUG[15225]: app_queue.c:500 changethread: Device 'IAX2/606' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.

[Mar 10 12:02:41] 
<-- SIP read from 192.168.1.86:5062: 


[Mar 10 12:02:41] --- (0 headers 0 lines) Nat keepalive ---

[Mar 10 12:02:41] 
<-- SIP read from 192.168.1.165:50206: 
REGISTER sip:24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK20fc6540
From: <sip:512@24.123.23.170;user=phone>
To: <sip:512@24.123.23.170;user=phone>
Call-ID: 000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165
CSeq: 11589 REGISTER
User-Agent: CSCO/7
Contact: <sip:512@192.168.1.165:5060>
Authorization: Digest username="512",realm="asterisk",uri="sip:24.123.23.170",response="b0c7cc5970bd5bc1d6ce2e497086c177",nonce="30041a4b",algorithm=MD5
Content-Length: 0
Expires: 60


[Mar 10 12:02:41] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34)
[Mar 10 12:02:41] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK20fc6540 (58)
[Mar 10 12:02:41] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: From: <sip:512@24.123.23.170;user=phone> (40)
[Mar 10 12:02:41] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: To: <sip:512@24.123.23.170;user=phone> (38)
[Mar 10 12:02:41] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Call-ID: 000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165 (58)
[Mar 10 12:02:41] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: CSeq: 11589 REGISTER (20)
[Mar 10 12:02:41] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: User-Agent: CSCO/7 (18)
[Mar 10 12:02:41] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Contact: <sip:512@192.168.1.165:5060> (37)
[Mar 10 12:02:41] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Authorization: Digest username="512",realm="asterisk",uri="sip:24.123.23.170",response="b0c7cc5970bd5bc1d6ce2e497086c177",nonce="30041a4b",algorithm=MD5 (152)
[Mar 10 12:02:41] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Content-Length: 0 (17)
[Mar 10 12:02:41] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Expires: 60 (11)
[Mar 10 12:02:41] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11:  (0)

[Mar 10 12:02:41] --- (11 headers 0 lines) ---
[Mar 10 12:02:41] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:02:41] Using latest REGISTER request as basis request

[Mar 10 12:02:41] Sending to 192.168.1.165 : 5060 (non-NAT)

[Mar 10 12:02:41] Transmitting (no NAT) to 192.168.1.165:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK20fc6540;received=192.168.1.165
From: <sip:512@24.123.23.170;user=phone>
To: <sip:512@24.123.23.170;user=phone>
Call-ID: 000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165
CSeq: 11589 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:512@24.123.23.170>
Content-Length: 0


---

[Mar 10 12:02:41] Transmitting (no NAT) to 192.168.1.165:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK20fc6540;received=192.168.1.165
From: <sip:512@24.123.23.170;user=phone>
To: <sip:512@24.123.23.170;user=phone>;tag=as75433f9c
Call-ID: 000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165
CSeq: 11589 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 60
Contact: <sip:512@192.168.1.165:5060>;expires=60
Date: Sat, 10 Mar 2007 17:02:41 GMT
Content-Length: 0


---

[Mar 10 12:02:41] Scheduling destruction of call '000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165' in 15000 ms
[Mar 10 12:02:41] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 512
[Mar 10 12:02:41] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/512 - state 1 (Not in use)
[Mar 10 12:02:41] DEBUG[15226]: app_queue.c:500 changethread: Device 'SIP/512' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:02:41] DEBUG[15136]: chan_iax2.c:9448 iax2_devicestate: Checking device state for device 616
[Mar 10 12:02:41] DEBUG[15136]: chan_iax2.c:9456 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 616? addr=1159759178, defaddr=0 maxms=0, lastms=0
[Mar 10 12:02:41] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for IAX2/616 - state 1 (Not in use)
[Mar 10 12:02:41] DEBUG[15227]: app_queue.c:500 changethread: Device 'IAX2/616' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:02:41] DEBUG[15136]: chan_iax2.c:9448 iax2_devicestate: Checking device state for device 616
[Mar 10 12:02:41] DEBUG[15136]: chan_iax2.c:9456 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 616? addr=1159759178, defaddr=0 maxms=0, lastms=0
[Mar 10 12:02:41] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for IAX2/616 - state 1 (Not in use)
[Mar 10 12:02:41] DEBUG[15228]: app_queue.c:500 changethread: Device 'IAX2/616' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.

[Mar 10 12:02:42] 
<-- SIP read from 192.168.1.165:50209: 
REGISTER sip:24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK489a757e
From: <sip:515@24.123.23.170;user=phone>
To: <sip:515@24.123.23.170;user=phone>
Call-ID: 000ff78d-ebb20007-4ad25281-15520cf9@192.168.1.165
CSeq: 11594 REGISTER
User-Agent: CSCO/7
Contact: <sip:515@192.168.1.165:5060>
Content-Length: 0
Expires: 60


[Mar 10 12:02:42] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34)
[Mar 10 12:02:42] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK489a757e (58)
[Mar 10 12:02:42] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: From: <sip:515@24.123.23.170;user=phone> (40)
[Mar 10 12:02:42] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: To: <sip:515@24.123.23.170;user=phone> (38)
[Mar 10 12:02:42] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Call-ID: 000ff78d-ebb20007-4ad25281-15520cf9@192.168.1.165 (58)
[Mar 10 12:02:42] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: CSeq: 11594 REGISTER (20)
[Mar 10 12:02:42] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: User-Agent: CSCO/7 (18)
[Mar 10 12:02:42] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Contact: <sip:515@192.168.1.165:5060> (37)
[Mar 10 12:02:42] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Content-Length: 0 (17)
[Mar 10 12:02:42] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Expires: 60 (11)
[Mar 10 12:02:42] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10:  (0)

[Mar 10 12:02:42] --- (10 headers 0 lines) ---
[Mar 10 12:02:42] DEBUG[15149]: chan_sip.c:3168 sip_alloc: Allocating new SIP dialog for 000ff78d-ebb20007-4ad25281-15520cf9@192.168.1.165 - REGISTER (No RTP)
[Mar 10 12:02:42] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:02:42] Using latest REGISTER request as basis request

[Mar 10 12:02:42] Sending to 192.168.1.165 : 5060 (non-NAT)

[Mar 10 12:02:42] Transmitting (no NAT) to 192.168.1.165:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK489a757e;received=192.168.1.165
From: <sip:515@24.123.23.170;user=phone>
To: <sip:515@24.123.23.170;user=phone>
Call-ID: 000ff78d-ebb20007-4ad25281-15520cf9@192.168.1.165
CSeq: 11594 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:515@24.123.23.170>
Content-Length: 0


---

[Mar 10 12:02:42] Transmitting (no NAT) to 192.168.1.165:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK489a757e;received=192.168.1.165
From: <sip:515@24.123.23.170;user=phone>
To: <sip:515@24.123.23.170;user=phone>;tag=as35d1fd38
Call-ID: 000ff78d-ebb20007-4ad25281-15520cf9@192.168.1.165
CSeq: 11594 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5d9c184d"
Content-Length: 0


---

[Mar 10 12:02:42] Scheduling destruction of call '000ff78d-ebb20007-4ad25281-15520cf9@192.168.1.165' in 15000 ms

[Mar 10 12:02:43] 
<-- SIP read from 192.168.1.165:50209: 
REGISTER sip:24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK674ab51c
From: <sip:515@24.123.23.170;user=phone>
To: <sip:515@24.123.23.170;user=phone>
Call-ID: 000ff78d-ebb20007-4ad25281-15520cf9@192.168.1.165
CSeq: 11595 REGISTER
User-Agent: CSCO/7
Contact: <sip:515@192.168.1.165:5060>
Authorization: Digest username="515",realm="asterisk",uri="sip:24.123.23.170",response="59299d08965df967834a99c8d3288e0e",nonce="5d9c184d",algorithm=MD5
Content-Length: 0
Expires: 60


[Mar 10 12:02:43] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34)
[Mar 10 12:02:43] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK674ab51c (58)
[Mar 10 12:02:43] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: From: <sip:515@24.123.23.170;user=phone> (40)
[Mar 10 12:02:43] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: To: <sip:515@24.123.23.170;user=phone> (38)
[Mar 10 12:02:43] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Call-ID: 000ff78d-ebb20007-4ad25281-15520cf9@192.168.1.165 (58)
[Mar 10 12:02:43] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: CSeq: 11595 REGISTER (20)
[Mar 10 12:02:43] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: User-Agent: CSCO/7 (18)
[Mar 10 12:02:43] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Contact: <sip:515@192.168.1.165:5060> (37)
[Mar 10 12:02:43] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Authorization: Digest username="515",realm="asterisk",uri="sip:24.123.23.170",response="59299d08965df967834a99c8d3288e0e",nonce="5d9c184d",algorithm=MD5 (152)
[Mar 10 12:02:43] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Content-Length: 0 (17)
[Mar 10 12:02:43] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Expires: 60 (11)
[Mar 10 12:02:43] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11:  (0)

[Mar 10 12:02:43] --- (11 headers 0 lines) ---
[Mar 10 12:02:43] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER

[Mar 10 12:02:43] Using latest REGISTER request as basis request

[Mar 10 12:02:43] Sending to 192.168.1.165 : 5060 (non-NAT)

[Mar 10 12:02:43] Transmitting (no NAT) to 192.168.1.165:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK674ab51c;received=192.168.1.165
From: <sip:515@24.123.23.170;user=phone>
To: <sip:515@24.123.23.170;user=phone>
Call-ID: 000ff78d-ebb20007-4ad25281-15520cf9@192.168.1.165
CSeq: 11595 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:515@24.123.23.170>
Content-Length: 0


---

[Mar 10 12:02:43] Transmitting (no NAT) to 192.168.1.165:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK674ab51c;received=192.168.1.165
From: <sip:515@24.123.23.170;user=phone>
To: <sip:515@24.123.23.170;user=phone>;tag=as35d1fd38
Call-ID: 000ff78d-ebb20007-4ad25281-15520cf9@192.168.1.165
CSeq: 11595 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 60
Contact: <sip:515@192.168.1.165:5060>;expires=60
Date: Sat, 10 Mar 2007 17:02:43 GMT
Content-Length: 0


---

[Mar 10 12:02:43] Scheduling destruction of call '000ff78d-ebb20007-4ad25281-15520cf9@192.168.1.165' in 15000 ms
[Mar 10 12:02:43] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 515
[Mar 10 12:02:43] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/515 - state 1 (Not in use)
[Mar 10 12:02:43] DEBUG[15229]: app_queue.c:500 changethread: Device 'SIP/515' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:02:44] DEBUG[15149]: chan_sip.c:1334 __sip_autodestruct: Auto destroying call '55ae66476793@24.123.23.170'

[Mar 10 12:02:44] Destroying call '55ae66476793@24.123.23.170'

[Mar 10 12:02:47] 
<-- SIP read from 192.168.1.83:5060: 
BYE sip:523@24.123.23.170 SIP/2.0
Max-Forwards: 70
Content-Length: 0
Via: SIP/2.0/UDP 192.168.1.83:5060;branch=z9hG4bK0c6d3696c
Call-ID: 20a99672af308312d36264481f6b7f5a@192.168.1.83
From: 593 <sip:593@24.123.23.170:5060>;tag=b3eeb4c9225baf5
To: 523 <sip:523@24.123.23.170:5060>;tag=as38239579
CSeq: 75546812 BYE
Supported: timer
Supported: replaces
Proxy-Authorization:Digest response="dfc4bd595350abb7ad91c8739ff3d136",username="593",realm="asterisk",nonce="4de95353",algorithm=MD5,uri="sip:523@24.123.23.170"
User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4


[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: BYE sip:523@24.123.23.170 SIP/2.0 (33)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: Max-Forwards: 70 (16)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: Content-Length: 0 (17)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: Via: SIP/2.0/UDP 192.168.1.83:5060;branch=z9hG4bK0c6d3696c (58)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Call-ID: 20a99672af308312d36264481f6b7f5a@192.168.1.83 (54)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: From: 593 <sip:593@24.123.23.170:5060>;tag=b3eeb4c9225baf5 (58)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: To: 523 <sip:523@24.123.23.170:5060>;tag=as38239579 (51)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: CSeq: 75546812 BYE (18)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Supported: timer (16)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Supported: replaces (19)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Proxy-Authorization:Digest response="dfc4bd595350abb7ad91c8739ff3d136",username="593",realm="asterisk",nonce="4de95353",algorithm=MD5,uri="sip:523@24.123.23.170" (161)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11: User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 (42)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 12:  (0)

[Mar 10 12:02:47] --- (12 headers 0 lines) ---
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:11225 handle_request: **** Received BYE (8) - Command in SIP BYE

[Mar 10 12:02:47] Sending to 192.168.1.83 : 5060 (non-NAT)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:10876 handle_request_bye: Received bye, issuing owner hangup
.
[Mar 10 12:02:47] Transmitting (no NAT) to 192.168.1.83:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.83:5060;branch=z9hG4bK0c6d3696c;received=192.168.1.83
From: 593 <sip:593@24.123.23.170:5060>;tag=b3eeb4c9225baf5
To: 523 <sip:523@24.123.23.170:5060>;tag=as38239579
Call-ID: 20a99672af308312d36264481f6b7f5a@192.168.1.83
CSeq: 75546812 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:523@24.123.23.170>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


---
[Mar 10 12:02:47] DEBUG[15195]: rtp.c:1718 ast_rtp_bridge: Oooh, got a hangup
[Mar 10 12:02:47] DEBUG[15195]: chan_sip.c:13028 sip_set_rtp_peer: Sending reinvite on SIP '5d28f46f54fd450475a9f02419f99674@24.123.23.170' - It's audio soon redirected to IP 24.123.23.170

[Mar 10 12:02:47] set_destination: Parsing <sip:523@192.168.1.85:5060> for address/port to send to

[Mar 10 12:02:47] set_destination: set destination to 192.168.1.85, port 5060

[Mar 10 12:02:47] We're at 24.123.23.170 port 17986

[Mar 10 12:02:47] Video is at 24.123.23.170 port 12600

[Mar 10 12:02:47] Adding codec 0x4 (ulaw) to SDP

[Mar 10 12:02:47] Adding codec 0x8 (alaw) to SDP

[Mar 10 12:02:47] Adding non-codec 0x1 (telephone-event) to SDP
[Mar 10 12:02:47] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 0: INVITE sip:523@192.168.1.85:5060 SIP/2.0 (40)
[Mar 10 12:02:47] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK51b2e827 (58)
[Mar 10 12:02:47] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 2: From: <sip:593@24.123.23.170>;tag=as0da2dd59 (44)
[Mar 10 12:02:47] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 3: To: 523 <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f (72)
[Mar 10 12:02:47] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 4: Contact: <sip:593@24.123.23.170> (32)
[Mar 10 12:02:47] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 5: Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170 (55)
[Mar 10 12:02:47] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 6: CSeq: 104 INVITE (16)
[Mar 10 12:02:47] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 7: User-Agent: Asterisk PBX (24)
[Mar 10 12:02:47] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 8: Max-Forwards: 70 (16)
[Mar 10 12:02:47] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66)
[Mar 10 12:02:47] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 10: X-asterisk-info: SIP re-invite (RTP bridge) (43)
[Mar 10 12:02:47] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 11: Content-Type: application/sdp (29)
[Mar 10 12:02:47] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 12: Content-Length: 242 (19)
[Mar 10 12:02:47] DEBUG[15195]: chan_sip.c:3392 parse_request: Header 13:  (0)
[Mar 10 12:02:47] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: v=0 (3)
[Mar 10 12:02:47] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: o=root 15131 15136 IN IP4 24.123.23.170 (39)
[Mar 10 12:02:47] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: s=session (9)
[Mar 10 12:02:47] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: c=IN IP4 24.123.23.170 (22)
[Mar 10 12:02:47] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: t=0 0 (5)
[Mar 10 12:02:47] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: m=audio 17986 RTP/AVP 0 8 101 (29)
[Mar 10 12:02:47] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: a=rtpmap:0 PCMU/8000 (20)
[Mar 10 12:02:47] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: a=rtpmap:8 PCMA/8000 (20)
[Mar 10 12:02:47] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33)
[Mar 10 12:02:47] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: a=fmtp:101 0-16 (15)
[Mar 10 12:02:47] DEBUG[15195]: chan_sip.c:3424 parse_request: Line: a=silenceSupp:off - - - - (25)

[Mar 10 12:02:47] 13 headers, 11 lines

[Mar 10 12:02:47] Reliably Transmitting (no NAT) to 192.168.1.85:5060:
INVITE sip:523@192.168.1.85:5060 SIP/2.0
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK51b2e827
From: <sip:593@24.123.23.170>;tag=as0da2dd59
To: 523 <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f
Contact: <sip:593@24.123.23.170>
Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 242

v=0
o=root 15131 15136 IN IP4 24.123.23.170
s=session
c=IN IP4 24.123.23.170
t=0 0
m=audio 17986 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
[Mar 10 12:02:47] DEBUG[15195]: chan_sip.c:1304 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id  #184
[Mar 10 12:02:47] DEBUG[15195]: channel.c:3600 ast_channel_bridge: Returning from native bridge, channels: SIP/593-084f44b8, SIP/523-0850b038
[Mar 10 12:02:47] DEBUG[15195]: channel.c:1373 ast_hangup: Hanging up channel 'SIP/523-0850b038'
[Mar 10 12:02:47] DEBUG[15195]: chan_sip.c:2427 sip_hangup: Hangup call SIP/523-0850b038, SIP callid 5d28f46f54fd450475a9f02419f99674@24.123.23.170)
[Mar 10 12:02:47] DEBUG[15195]: chan_sip.c:2435 sip_hangup: update_call_counter(523) - decrement call limit counter
[Mar 10 12:02:47] DEBUG[15195]: chan_sip.c:2217 update_call_counter: Updating call counter for outgoing call

[Mar 10 12:02:47] Scheduling destruction of call '5d28f46f54fd450475a9f02419f99674@24.123.23.170' in 32000 ms
[Mar 10 12:02:47] DEBUG[15195]: app_dial.c:1635 dial_exec_full: Exiting with DIALSTATUS=ANSWER.
[Mar 10 12:02:47] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 523
[Mar 10 12:02:47] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/523 - state 1 (Not in use)
[Mar 10 12:02:47] DEBUG[15195]: pbx.c:2316 __ast_pbx_run: Spawn extension (smvoice-sip,523,6) exited non-zero on 'SIP/593-084f44b8'

[Mar 10 12:02:47]   == Spawn extension (smvoice-sip, 523, 6) exited non-zero on 'SIP/593-084f44b8'
[Mar 10 12:02:47] DEBUG[15195]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '"593 593" <593>'
[Mar 10 12:02:47] DEBUG[15195]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '593'
[Mar 10 12:02:47] DEBUG[15195]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '523'
[Mar 10 12:02:47] DEBUG[15195]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'smvoice-sip'
[Mar 10 12:02:47] DEBUG[15195]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'SIP/593-084f44b8'
[Mar 10 12:02:47] DEBUG[15195]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'SIP/523-0850b038'
[Mar 10 12:02:47] DEBUG[15195]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'Dial'
[Mar 10 12:02:47] DEBUG[15195]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'SIP/523|20|'
[Mar 10 12:02:47] DEBUG[15195]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '2007-03-10 12:02:02'
[Mar 10 12:02:47] DEBUG[15195]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '2007-03-10 12:02:05'
[Mar 10 12:02:47] DEBUG[15195]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '2007-03-10 12:02:47'
[Mar 10 12:02:47] DEBUG[15195]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '45'
[Mar 10 12:02:47] DEBUG[15195]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '42'
[Mar 10 12:02:47] DEBUG[15195]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'ANSWERED'
[Mar 10 12:02:47] DEBUG[15195]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION'
[Mar 10 12:02:47] DEBUG[15195]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '(null)'
[Mar 10 12:02:47] DEBUG[15195]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '1173546122.0'
[Mar 10 12:02:47] DEBUG[15195]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '(null)'
[Mar 10 12:02:47] DEBUG[15195]: channel.c:1373 ast_hangup: Hanging up channel 'SIP/593-084f44b8'
[Mar 10 12:02:47] DEBUG[15195]: chan_sip.c:2427 sip_hangup: Hangup call SIP/593-084f44b8, SIP callid 20a99672af308312d36264481f6b7f5a@192.168.1.83)
[Mar 10 12:02:47] DEBUG[15195]: chan_sip.c:2435 sip_hangup: update_call_counter(593) - decrement call limit counter
[Mar 10 12:02:47] DEBUG[15195]: chan_sip.c:2217 update_call_counter: Updating call counter for outgoing call
[Mar 10 12:02:47] DEBUG[15136]: chan_sip.c:11767 sip_devicestate: Checking device state for peer 593
[Mar 10 12:02:47] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for SIP/593 - state 1 (Not in use)
[Mar 10 12:02:47] DEBUG[15230]: app_queue.c:500 changethread: Device 'SIP/523' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:02:47] DEBUG[15231]: app_queue.c:500 changethread: Device 'SIP/593' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.

[Mar 10 12:02:47] 
<-- SIP read from 192.168.1.85:5060: 
SIP/2.0 200 OK
To: 523 <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f
Contact: <sip:523@192.168.1.85:5060>
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK51b2e827
From: <sip:593@24.123.23.170>;tag=as0da2dd59
Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170
CSeq: 104 INVITE
Content-Type: application/sdp
Content-Length: 223
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK

v=0
o=- 15131 248059 IN IP4 192.168.1.85
s=session
c=IN IP4 192.168.1.85
t=0 0
m=audio 30012 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,16,17,18
a=silenceSupp:off - - - -

[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: SIP/2.0 200 OK (14)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: To: 523 <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f (72)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: Contact: <sip:523@192.168.1.85:5060> (36)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK51b2e827 (58)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: From: <sip:593@24.123.23.170>;tag=as0da2dd59 (44)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170 (55)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: CSeq: 104 INVITE (16)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Content-Type: application/sdp (29)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Content-Length: 223 (19)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9: Supported: sip-cc, sip-cc-01, replaces, timer (45)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 11:  (0)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: v=0 (3)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: o=- 15131 248059 IN IP4 192.168.1.85 (36)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: s=session (9)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: c=IN IP4 192.168.1.85 (21)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: t=0 0 (5)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: m=audio 30012 RTP/AVP 0 101 (27)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:0 PCMU/8000 (20)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=fmtp:101 0-15,16,17,18 (24)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3424 parse_request: Line: a=silenceSupp:off - - - - (25)

[Mar 10 12:02:47] --- (11 headers 10 lines) ---
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:1390 __sip_ack: Acked pending invite 104
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:1401 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #184
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:1412 __sip_ack: Stopping retransmission on '5d28f46f54fd450475a9f02419f99674@24.123.23.170' of Request 104: Match Found
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:9642 handle_response_invite: SIP response 200 to standard invite

[Mar 10 12:02:47] Found RTP audio format 0

[Mar 10 12:02:47] Found RTP audio format 101

[Mar 10 12:02:47] Peer audio RTP is at port 192.168.1.85:30012
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3631 process_sdp: Peer audio RTP is at port 192.168.1.85:30012

[Mar 10 12:02:47] Peer video RTP is at port 192.168.1.85:65535
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3653 process_sdp: Peer video RTP is at port 192.168.1.85:65535

[Mar 10 12:02:47] Found description format PCMU

[Mar 10 12:02:47] Found description format telephone-event

[Mar 10 12:02:47] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)

[Mar 10 12:02:47] Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:6137 build_route: build_route: Retaining previous route: <sip:523@192.168.1.85:5060>

[Mar 10 12:02:47] set_destination: Parsing <sip:523@192.168.1.85:5060> for address/port to send to

[Mar 10 12:02:47] set_destination: set destination to 192.168.1.85, port 5060

[Mar 10 12:02:47] Transmitting (no NAT) to 192.168.1.85:5060:
ACK sip:523@192.168.1.85:5060 SIP/2.0
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK45b76c66
From: <sip:593@24.123.23.170>;tag=as0da2dd59
To: 523 <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f
Contact: <sip:593@24.123.23.170>
Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170
CSeq: 104 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---

[Mar 10 12:02:47] set_destination: Parsing <sip:523@192.168.1.85:5060> for address/port to send to

[Mar 10 12:02:47] set_destination: set destination to 192.168.1.85, port 5060

[Mar 10 12:02:47] Reliably Transmitting (no NAT) to 192.168.1.85:5060:
BYE sip:523@192.168.1.85:5060 SIP/2.0
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK4414cde5
From: <sip:593@24.123.23.170>;tag=as0da2dd59
To: 523 <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f
Contact: <sip:593@24.123.23.170>
Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170
CSeq: 105 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:1304 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id  #186

[Mar 10 12:02:47] Scheduling destruction of call '5d28f46f54fd450475a9f02419f99674@24.123.23.170' in 32000 ms

[Mar 10 12:02:47] Destroying call '20a99672af308312d36264481f6b7f5a@192.168.1.83'

[Mar 10 12:02:47] 
<-- SIP read from 192.168.1.85:5060: 
SIP/2.0 200 OK
To: 523 <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK4414cde5
From: <sip:593@24.123.23.170>;tag=as0da2dd59
Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170
CSeq: 105 BYE
Content-Length: 0
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK


[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0: SIP/2.0 200 OK (14)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 1: To: 523 <sip:523@192.168.1.85:5060>;tag=632bf8b74949ce01359fc10b9b0b312f (72)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 2: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK4414cde5 (58)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 3: From: <sip:593@24.123.23.170>;tag=as0da2dd59 (44)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 4: Call-ID: 5d28f46f54fd450475a9f02419f99674@24.123.23.170 (55)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 5: CSeq: 105 BYE (13)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 6: Content-Length: 0 (17)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 7: Supported: sip-cc, sip-cc-01, replaces, timer (45)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 8: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62)
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 9:  (0)

[Mar 10 12:02:47] --- (9 headers 0 lines) ---
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:1401 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #186
[Mar 10 12:02:47] DEBUG[15149]: chan_sip.c:1412 __sip_ack: Stopping retransmission on '5d28f46f54fd450475a9f02419f99674@24.123.23.170' of Request 105: Match Found

[Mar 10 12:02:47] Destroying call '5d28f46f54fd450475a9f02419f99674@24.123.23.170'
[Mar 10 12:02:48] DEBUG[15149]: chan_sip.c:1334 __sip_autodestruct: Auto destroying call '55ae6647678c@24.123.23.170'

[Mar 10 12:02:48] Destroying call '55ae6647678c@24.123.23.170'
[Mar 10 12:02:49] DEBUG[15136]: chan_iax2.c:9448 iax2_devicestate: Checking device state for device 614
[Mar 10 12:02:49] DEBUG[15136]: chan_iax2.c:9456 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 614? addr=-1493063488, defaddr=0 maxms=0, lastms=0
[Mar 10 12:02:49] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for IAX2/614 - state 1 (Not in use)
[Mar 10 12:02:49] DEBUG[15232]: app_queue.c:500 changethread: Device 'IAX2/614' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:02:49] DEBUG[15136]: chan_iax2.c:9448 iax2_devicestate: Checking device state for device 614
[Mar 10 12:02:49] DEBUG[15136]: chan_iax2.c:9456 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 614? addr=-1493063488, defaddr=0 maxms=0, lastms=0
[Mar 10 12:02:49] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for IAX2/614 - state 1 (Not in use)
[Mar 10 12:02:49] DEBUG[15233]: app_queue.c:500 changethread: Device 'IAX2/614' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.

[Mar 10 12:02:49] 
<-- SIP read from 192.168.1.97:5060: 


[Mar 10 12:02:49] DEBUG[15149]: chan_sip.c:3392 parse_request: Header 0:  (0)

[Mar 10 12:02:49] --- (0 headers 0 lines) Nat keepalive ---
[Mar 10 12:02:50] DEBUG[15136]: chan_iax2.c:9448 iax2_devicestate: Checking device state for device 651
[Mar 10 12:02:50] DEBUG[15136]: chan_iax2.c:9456 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 651? addr=-1459509056, defaddr=0 maxms=0, lastms=0
[Mar 10 12:02:50] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for IAX2/651 - state 1 (Not in use)
[Mar 10 12:02:50] DEBUG[15234]: app_queue.c:500 changethread: Device 'IAX2/651' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:02:50] DEBUG[15136]: chan_iax2.c:9448 iax2_devicestate: Checking device state for device 651
[Mar 10 12:02:50] DEBUG[15136]: chan_iax2.c:9456 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 651? addr=-1459509056, defaddr=0 maxms=0, lastms=0
[Mar 10 12:02:50] DEBUG[15136]: devicestate.c:187 do_state_change: Changing state for IAX2/651 - state 1 (Not in use)
[Mar 10 12:02:50] DEBUG[15235]: app_queue.c:500 changethread: Device 'IAX2/651' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 12:02:50] DEBUG[15149]: chan_sip.c:1334 __sip_autodestruct: Auto destroying call '55ae66469f11@24.123.23.170'

[Mar 10 12:02:50] Destroying call '55ae66469f11@24.123.23.170'
s
[Mar 10 12:02:51] 
<-- SIP read from 192.168.1.86:5060: 


[Mar 10 12:02:51] --- (0 headers 0 lines) Nat keepalive ---
[Mar 10 12:02:51] DEBUG[15149]: chan_sip.c:1334 __sip_autodestruct: Auto destroying call '55ae6647666e@24.123.23.170'

[Mar 10 12:02:51] Destroying call '55ae6647666e@24.123.23.170'
top now[Mar 10 12:02:52] DEBUG[15149]: chan_sip.c:1334 __sip_autodestruct: Auto destroying call '000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165'

[Mar 10 12:02:52] Destroying call '000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165'


[Mar 10 12:02:53] Beginning asterisk shutdown....

[Mar 10 12:02:53] Executing last minute cleanups

[Mar 10 12:02:53]   == Destroying musiconhold processes
[Mar 10 12:02:53] DEBUG[15131]: res_musiconhold.c:1091 ast_moh_destroy: killing 15139!
[Mar 10 12:02:53] DEBUG[15131]: res_musiconhold.c:1106 ast_moh_destroy: mpg123 pid 15139 and child died after 2088634 bytes read

[Mar 10 12:02:53] Asterisk cleanly ending (0).
[Mar 10 12:02:53] DEBUG[15131]: asterisk.c:900 quit_handler: Asterisk ending (0).