[Feb 22 17:22:17] Asterisk 1.4.0, Copyright (C) 1999 - 2006 Digium, Inc. and others. [Feb 22 17:22:17] Created by Mark Spencer [Feb 22 17:22:17] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. [Feb 22 17:22:17] This is free software, with components licensed under the GNU General Public [Feb 22 17:22:17] License version 2 and other licenses; you are welcome to redistribute it under [Feb 22 17:22:17] certain conditions. Type 'core show license' for details. [Feb 22 17:22:17] ========================================================================= [Feb 22 17:22:17] == Parsing '/etc/asterisk/asterisk.conf': Parsing /etc/asterisk/asterisk.conf [Feb 22 17:22:17] Found [Feb 22 17:22:17] == Parsing '/etc/asterisk/extconfig.conf': Parsing /etc/asterisk/extconfig.conf [Feb 22 17:22:17] Found [Feb 22 17:22:17] == Parsing '/etc/asterisk/logger.conf': Parsing /etc/asterisk/logger.conf [Feb 22 17:22:17] Found [Feb 22 17:22:17] Asterisk Event Logger Started /var/log/asterisk/event_log [Feb 22 17:22:17] Asterisk Dynamic Loader Starting: [Feb 22 17:22:17] == Parsing '/etc/asterisk/modules.conf': [Feb 22 17:22:17] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/modules.conf [Feb 22 17:22:17] Found [Feb 22 17:22:17] == Parsing '/etc/asterisk/dnsmgr.conf': [Feb 22 17:22:17] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/dnsmgr.conf [Feb 22 17:22:17] Found [Feb 22 17:22:17] == Manager registered action Ping [Feb 22 17:22:17] == Manager registered action Events [Feb 22 17:22:17] == Manager registered action Logoff [Feb 22 17:22:17] == Manager registered action Hangup [Feb 22 17:22:17] == Manager registered action Status [Feb 22 17:22:17] == Manager registered action Setvar [Feb 22 17:22:17] == Manager registered action Getvar [Feb 22 17:22:17] == Manager registered action GetConfig [Feb 22 17:22:17] == Manager registered action UpdateConfig [Feb 22 17:22:17] == Manager registered action Redirect [Feb 22 17:22:17] == Manager registered action Originate [Feb 22 17:22:17] == Manager registered action Command [Feb 22 17:22:17] == Manager registered action ExtensionState [Feb 22 17:22:17] == Manager registered action AbsoluteTimeout [Feb 22 17:22:17] == Manager registered action MailboxStatus [Feb 22 17:22:17] == Manager registered action MailboxCount [Feb 22 17:22:17] == Manager registered action ListCommands [Feb 22 17:22:17] == Manager registered action UserEvent [Feb 22 17:22:17] == Manager registered action WaitEvent [Feb 22 17:22:17] == Parsing '/etc/asterisk/manager.conf': [Feb 22 17:22:17] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/manager.conf [Feb 22 17:22:17] Found [Feb 22 17:22:17] Asterisk Management interface listening on port 5038 [Feb 22 17:22:17] == Parsing '/etc/asterisk/cdr.conf': [Feb 22 17:22:17] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/cdr.conf [Feb 22 17:22:17] Found [Feb 22 17:22:17] NOTICE[14736]: cdr.c:1092 do_reload: CDR simple logging enabled. [Feb 22 17:22:17] == Parsing '/etc/asterisk/rtp.conf': [Feb 22 17:22:17] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/rtp.conf [Feb 22 17:22:17] Found [Feb 22 17:22:17] == RTP Allocating from port range 10000 -> 20000 [Feb 22 17:22:17] == UDPTL allocating from port range 4500 -> 4999 [Feb 22 17:22:17] Asterisk PBX Core Initializing [Feb 22 17:22:17] Registering builtin applications: [Feb 22 17:22:17] [Answer] [Feb 22 17:22:17] == Registered application 'Answer' [Feb 22 17:22:17] [BackGround] [Feb 22 17:22:17] == Registered application 'BackGround' [Feb 22 17:22:17] [Busy] [Feb 22 17:22:17] == Registered application 'Busy' [Feb 22 17:22:17] [Congestion] [Feb 22 17:22:17] == Registered application 'Congestion' [Feb 22 17:22:17] [Goto] [Feb 22 17:22:17] == Registered application 'Goto' [Feb 22 17:22:17] [GotoIf] [Feb 22 17:22:17] == Registered application 'GotoIf' [Feb 22 17:22:17] [GotoIfTime] [Feb 22 17:22:17] == Registered application 'GotoIfTime' [Feb 22 17:22:17] [ExecIfTime] [Feb 22 17:22:17] == Registered application 'ExecIfTime' [Feb 22 17:22:17] [Hangup] [Feb 22 17:22:17] == Registered application 'Hangup' [Feb 22 17:22:17] [NoOp] [Feb 22 17:22:17] == Registered application 'NoOp' [Feb 22 17:22:17] [Progress] [Feb 22 17:22:17] == Registered application 'Progress' [Feb 22 17:22:17] [ResetCDR] [Feb 22 17:22:17] == Registered application 'ResetCDR' [Feb 22 17:22:17] [Ringing] [Feb 22 17:22:17] == Registered application 'Ringing' [Feb 22 17:22:17] [SayNumber] [Feb 22 17:22:17] == Registered application 'SayNumber' [Feb 22 17:22:17] [SayDigits] [Feb 22 17:22:17] == Registered application 'SayDigits' [Feb 22 17:22:17] [SayAlpha] [Feb 22 17:22:17] == Registered application 'SayAlpha' [Feb 22 17:22:17] [SayPhonetic] [Feb 22 17:22:17] == Registered application 'SayPhonetic' [Feb 22 17:22:17] [SetAMAFlags] [Feb 22 17:22:17] == Registered application 'SetAMAFlags' [Feb 22 17:22:17] [SetGlobalVar] [Feb 22 17:22:17] == Registered application 'SetGlobalVar' [Feb 22 17:22:17] [Set] [Feb 22 17:22:17] == Registered application 'Set' [Feb 22 17:22:17] [ImportVar] [Feb 22 17:22:17] == Registered application 'ImportVar' [Feb 22 17:22:17] [Wait] [Feb 22 17:22:17] == Registered application 'Wait' [Feb 22 17:22:17] [WaitExten] [Feb 22 17:22:17] == Registered application 'WaitExten' [Feb 22 17:22:17] == Manager registered action DBGet [Feb 22 17:22:17] == Manager registered action DBPut [Feb 22 17:22:17] == Parsing '/etc/asterisk/enum.conf': [Feb 22 17:22:17] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/enum.conf [Feb 22 17:22:17] Found [Feb 22 17:22:17] Asterisk Dynamic Loader Starting: [Feb 22 17:22:17] == Parsing '/etc/asterisk/modules.conf': [Feb 22 17:22:17] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/modules.conf [Feb 22 17:22:17] Found [Feb 22 17:22:17] NOTICE[14736]: loader.c:792 load_modules: 146 modules will be loaded. [Feb 22 17:22:17] == Registered application 'MusicOnHold' [Feb 22 17:22:17] == Registered application 'WaitMusicOnHold' [Feb 22 17:22:17] == Registered application 'SetMusicOnHold' [Feb 22 17:22:17] == Registered application 'StartMusicOnHold' [Feb 22 17:22:17] == Registered application 'StopMusicOnHold' [Feb 22 17:22:17] == Parsing '/etc/asterisk/musiconhold.conf': [Feb 22 17:22:17] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/musiconhold.conf [Feb 22 17:22:17] Found [Feb 22 17:22:17] WARNING[14736]: res_musiconhold.c:1049 load_moh_classes: The old musiconhold.conf syntax has been deprecated! Please refer to the sample configuration for information on the new syntax. [Feb 22 17:22:17] res_musiconhold.so => (Music On Hold Resource) [Feb 22 17:22:17] WARNING[14742]: res_musiconhold.c:351 spawn_mp3: /var/lib/asterisk/mohmp3.msgnet is not a valid directory [Feb 22 17:22:17] WARNING[14742]: res_musiconhold.c:502 monmp3thread: Unable to spawn mp3player [Feb 22 17:22:17] == Registered application 'Monitor' [Feb 22 17:22:17] == Registered application 'StopMonitor' [Feb 22 17:22:17] == Registered application 'ChangeMonitor' [Feb 22 17:22:17] == Registered application 'PauseMonitor' [Feb 22 17:22:17] == Registered application 'UnpauseMonitor' [Feb 22 17:22:17] == Manager registered action Monitor [Feb 22 17:22:17] == Manager registered action StopMonitor [Feb 22 17:22:17] == Manager registered action ChangeMonitor [Feb 22 17:22:17] == Manager registered action PauseMonitor [Feb 22 17:22:17] == Manager registered action UnpauseMonitor [Feb 22 17:22:17] res_monitor.so => (Call Monitoring Resource) [Feb 22 17:22:17] == Parsing '/etc/asterisk/indications.conf': [Feb 22 17:22:17] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/indications.conf [Feb 22 17:22:17] Found [Feb 22 17:22:17] -- Registered indication country 'cl' [Feb 22 17:22:17] -- Registered indication country 'tw' [Feb 22 17:22:17] -- Registered indication country 'us' [Feb 22 17:22:17] -- Registered indication country 'au' [Feb 22 17:22:17] -- Registered indication country 'fr' [Feb 22 17:22:17] -- Registered indication country 'de' [Feb 22 17:22:17] -- Registered indication country 'nl' [Feb 22 17:22:17] -- Registered indication country 'uk' [Feb 22 17:22:17] -- Registered indication country 'fi' [Feb 22 17:22:17] -- Registered indication country 'no' [Feb 22 17:22:17] -- Registered indication country 'br' [Feb 22 17:22:17] -- Registered indication country 'za' [Feb 22 17:22:17] -- Registered indication country 'it' [Feb 22 17:22:17] -- Registered indication country 'us-o' [Feb 22 17:22:17] -- Registered indication country 'gr' [Feb 22 17:22:17] -- Registered indication country 'ru' [Feb 22 17:22:17] -- Registered indication country 'nz' [Feb 22 17:22:17] -- Setting default indication country to 'us' [Feb 22 17:22:17] == Registered application 'PlayTones' [Feb 22 17:22:17] == Registered application 'StopPlayTones' [Feb 22 17:22:17] res_indications.so => (Indications Resource) [Feb 22 17:22:18] == Parsing '/etc/asterisk/features.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/features.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] DEBUG[14736]: pbx.c:3824 __ast_context_create: Registered context 'parkedcalls' [Feb 22 17:22:18] -- Registered extension context 'parkedcalls' [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '700' priority 1 to parkedcalls [Feb 22 17:22:18] -- Added extension '700' priority 1 to parkedcalls [Feb 22 17:22:18] DEBUG[14736]: res_features.c:277 notify_metermaids: Notification of state change to metermaids 700@parkedcalls [Feb 22 17:22:18] DEBUG[14736]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel park:700@parkedcalls [Feb 22 17:22:18] DEBUG[14741]: devicestate.c:157 ast_device_state: Checking if I can find provider for "park" - number: 700@parkedcalls [Feb 22 17:22:18] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for park:700@parkedcalls - state 4 (Invalid) [Feb 22 17:22:18] == Registered application 'ParkedCall' [Feb 22 17:22:18] == Registered application 'Park' [Feb 22 17:22:18] == Manager registered action ParkedCalls [Feb 22 17:22:18] == Manager registered action Park [Feb 22 17:22:18] res_features.so => (Call Features Resource) [Feb 22 17:22:18] -- Loaded PUBLIC key 'freeworlddialup' [Feb 22 17:22:18] DEBUG[14736]: res_crypto.c:259 try_load_key: Key 'freeworlddialup' loaded OK [Feb 22 17:22:18] -- Loaded PUBLIC key 'iaxtel' [Feb 22 17:22:18] DEBUG[14736]: res_crypto.c:259 try_load_key: Key 'iaxtel' loaded OK [Feb 22 17:22:18] res_crypto.so => (Cryptographic Digital Signatures) [Feb 22 17:22:18] == Parsing '/etc/asterisk/adsi.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/adsi.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] res_adsi.so => (ADSI Resource (not optional)) [Feb 22 17:22:18] res_speech.so => (Generic Speech Recognition API) [Feb 22 17:22:18] NOTICE[14736]: res_smdi.c:539 smdi_load: Unable to load config smdi.conf: SMDI disabled [Feb 22 17:22:18] WARNING[14736]: res_smdi.c:722 load_module: No SMDI interfaces are available to listen on, not starting SDMI listener. [Feb 22 17:22:18] == Registered application 'SetCDRUserField' [Feb 22 17:22:18] == Registered application 'AppendCDRUserField' [Feb 22 17:22:18] == Manager registered action SetCDRUserField [Feb 22 17:22:18] app_setcdruserfield.so => (CDR user field apps) [Feb 22 17:22:18] == Parsing '/etc/asterisk/queues.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/queues.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] == Registered application 'Queue' [Feb 22 17:22:18] == Registered application 'AddQueueMember' [Feb 22 17:22:18] == Registered application 'RemoveQueueMember' [Feb 22 17:22:18] == Registered application 'PauseQueueMember' [Feb 22 17:22:18] == Registered application 'UnpauseQueueMember' [Feb 22 17:22:18] == Registered application 'QueueLog' [Feb 22 17:22:18] == Manager registered action Queues [Feb 22 17:22:18] == Manager registered action QueueStatus [Feb 22 17:22:18] == Manager registered action QueueAdd [Feb 22 17:22:18] == Manager registered action QueueRemove [Feb 22 17:22:18] == Manager registered action QueuePause [Feb 22 17:22:18] == Registered custom function QUEUEAGENTCOUNT [Feb 22 17:22:18] == Registered custom function QUEUE_MEMBER_COUNT [Feb 22 17:22:18] == Registered custom function QUEUE_MEMBER_LIST [Feb 22 17:22:18] == Registered custom function QUEUE_WAITING_COUNT [Feb 22 17:22:18] app_queue.so => (True Call Queueing) [Feb 22 17:22:18] pbx_spool.so => (Outgoing Spool Support) [Feb 22 17:22:18] == Registered application 'Dictate' [Feb 22 17:22:18] app_dictate.so => (Virtual Dictation Machine) [Feb 22 17:22:18] == Registered application 'DISA' [Feb 22 17:22:18] app_disa.so => (DISA (Direct Inward System Access) Application) [Feb 22 17:22:18] == Registered custom function DB [Feb 22 17:22:18] == Registered custom function DB_EXISTS [Feb 22 17:22:18] == Registered custom function DB_DELETE [Feb 22 17:22:18] func_db.so => (Database (astdb) related dialplan functions) [Feb 22 17:22:18] == Registered custom function MUSICCLASS [Feb 22 17:22:18] func_moh.so => (Music-on-hold dialplan function) [Feb 22 17:22:18] == Registered application 'ChanIsAvail' [Feb 22 17:22:18] app_chanisavail.so => (Check channel availability) [Feb 22 17:22:18] == Parsing '/etc/asterisk/cdr_custom.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/cdr_custom.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] cdr_custom.so => (Customizable Comma Separated Values CDR Backend) [Feb 22 17:22:18] == Parsing '/etc/asterisk/codecs.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/codecs.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] -- codec_alaw: using generic PLC [Feb 22 17:22:18] WARNING[14736]: translate.c:675 __ast_register_translator: plc_samples 160 format 6 [Feb 22 17:22:18] == Registered translator 'alawtolin' from format alaw to slin, cost 1 [Feb 22 17:22:18] DEBUG[14736]: translate.c:425 rebuild_matrix: Resetting translation matrix [Feb 22 17:22:18] == Registered translator 'lintoalaw' from format slin to alaw, cost 1 [Feb 22 17:22:18] DEBUG[14736]: translate.c:425 rebuild_matrix: Resetting translation matrix [Feb 22 17:22:18] codec_alaw.so => (A-law Coder/Decoder) [Feb 22 17:22:18] == Registered custom function MD5 [Feb 22 17:22:18] == Registered custom function CHECK_MD5 [Feb 22 17:22:18] func_md5.so => (MD5 digest dialplan functions) [Feb 22 17:22:18] == Registered custom function SHA1 [Feb 22 17:22:18] func_sha1.so => (SHA-1 computation dialplan function) [Feb 22 17:22:18] == Registered custom function GLOBAL [Feb 22 17:22:18] func_global.so => (Global variable dialplan functions) [Feb 22 17:22:18] == Registered file format ogg_vorbis, extension(s) ogg [Feb 22 17:22:18] format_ogg_vorbis.so => (OGG/Vorbis audio) [Feb 22 17:22:18] == Registered application 'ZapRAS' [Feb 22 17:22:18] app_zapras.so => (Zap RAS Application) [Feb 22 17:22:18] == Registered file format pcm, extension(s) pcm|ulaw|ul|mu [Feb 22 17:22:18] == Registered file format alaw, extension(s) alaw|al [Feb 22 17:22:18] == Registered file format au, extension(s) au [Feb 22 17:22:18] == Registered file format g722, extension(s) g722 [Feb 22 17:22:18] format_pcm.so => (Raw/Sun uLaw/ALaw 8KHz Audio support (PCM,PCMA,AU) and G.722 16Khz Audio Support) [Feb 22 17:22:18] == Registered file format g726-40, extension(s) g726-40 [Feb 22 17:22:18] == Registered file format g726-32, extension(s) g726-32 [Feb 22 17:22:18] == Registered file format g726-24, extension(s) g726-24 [Feb 22 17:22:18] == Registered file format g726-16, extension(s) g726-16 [Feb 22 17:22:18] format_g726.so => (Raw G.726 (16/24/32/40kbps) data) [Feb 22 17:22:18] == Registered custom function GROUP_COUNT [Feb 22 17:22:18] == Registered custom function GROUP_MATCH_COUNT [Feb 22 17:22:18] == Registered custom function GROUP_LIST [Feb 22 17:22:18] == Registered custom function GROUP [Feb 22 17:22:18] func_groupcount.so => (Channel group dialplan functions) [Feb 22 17:22:18] == Registered application 'Page' [Feb 22 17:22:18] app_page.so => (Page Multiple Phones) [Feb 22 17:22:18] == Registered application 'Random' [Feb 22 17:22:18] app_random.so => (Random goto) [Feb 22 17:22:18] == Registered file format gsm, extension(s) gsm [Feb 22 17:22:18] format_gsm.so => (Raw GSM data) [Feb 22 17:22:18] == Registered translator 'ilbctolin' from format ilbc to slin, cost 2 [Feb 22 17:22:18] DEBUG[14736]: translate.c:425 rebuild_matrix: Resetting translation matrix [Feb 22 17:22:18] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:18] == Registered translator 'lintoilbc' from format slin to ilbc, cost 14 [Feb 22 17:22:18] DEBUG[14736]: translate.c:425 rebuild_matrix: Resetting translation matrix [Feb 22 17:22:18] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Feb 22 17:22:18] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:18] codec_ilbc.so => (iLBC Coder/Decoder) [Feb 22 17:22:18] NOTICE[14736]: pbx_ael.c:3859 pbx_load_module: Starting AEL load process. [Feb 22 17:22:18] NOTICE[14736]: pbx_ael.c:3866 pbx_load_module: AEL load process: calculated config file name '/etc/asterisk/extensions.ael'. [Feb 22 17:22:18] NOTICE[14736]: pbx_ael.c:3874 pbx_load_module: AEL load process: parsed config file name '/etc/asterisk/extensions.ael'. [Feb 22 17:22:18] NOTICE[14736]: pbx_ael.c:3877 pbx_load_module: AEL load process: checked config file name '/etc/asterisk/extensions.ael'. [Feb 22 17:22:18] DEBUG[14736]: pbx.c:3824 __ast_context_create: Registered context 'macro-std-exten-ael' [Feb 22 17:22:18] -- Registered extension context 'macro-std-exten-ael' [Feb 22 17:22:18] DEBUG[14736]: pbx.c:3824 __ast_context_create: Registered context 'ael-demo' [Feb 22 17:22:18] -- Registered extension context 'ael-demo' [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 1 to macro-std-exten-ael [Feb 22 17:22:18] -- Added extension 's' priority 1 to macro-std-exten-ael [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 2 to macro-std-exten-ael [Feb 22 17:22:18] -- Added extension 's' priority 2 to macro-std-exten-ael [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 3 to macro-std-exten-ael [Feb 22 17:22:18] -- Added extension 's' priority 3 to macro-std-exten-ael [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 4 to macro-std-exten-ael [Feb 22 17:22:18] -- Added extension 's' priority 4 to macro-std-exten-ael [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 5 to macro-std-exten-ael [Feb 22 17:22:18] -- Added extension 's' priority 5 to macro-std-exten-ael [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'a' priority 1 to macro-std-exten-ael [Feb 22 17:22:18] -- Added extension 'a' priority 1 to macro-std-exten-ael [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'a' priority 2 to macro-std-exten-ael [Feb 22 17:22:18] -- Added extension 'a' priority 2 to macro-std-exten-ael [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'a' priority 3 to macro-std-exten-ael [Feb 22 17:22:18] -- Added extension 'a' priority 3 to macro-std-exten-ael [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_sw-1-.' priority 1 to macro-std-exten-ael [Feb 22 17:22:18] -- Added extension '_sw-1-.' priority 1 to macro-std-exten-ael [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_sw-1-.' priority 2 to macro-std-exten-ael [Feb 22 17:22:18] -- Added extension '_sw-1-.' priority 2 to macro-std-exten-ael [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'sw-1-BUSY' priority 1 to macro-std-exten-ael [Feb 22 17:22:18] -- Added extension 'sw-1-BUSY' priority 1 to macro-std-exten-ael [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'sw-1-BUSY' priority 2 to macro-std-exten-ael [Feb 22 17:22:18] -- Added extension 'sw-1-BUSY' priority 2 to macro-std-exten-ael [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 1 to ael-demo [Feb 22 17:22:18] -- Added extension 's' priority 1 to ael-demo [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 2 to ael-demo [Feb 22 17:22:18] -- Added extension 's' priority 2 to ael-demo [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 3 to ael-demo [Feb 22 17:22:18] -- Added extension 's' priority 3 to ael-demo [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 4 to ael-demo [Feb 22 17:22:18] -- Added extension 's' priority 4 to ael-demo [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 5 to ael-demo [Feb 22 17:22:18] -- Added extension 's' priority 5 to ael-demo [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 6 to ael-demo [Feb 22 17:22:18] -- Added extension 's' priority 6 to ael-demo [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 7 to ael-demo [Feb 22 17:22:18] -- Added extension 's' priority 7 to ael-demo [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 8 to ael-demo [Feb 22 17:22:18] -- Added extension 's' priority 8 to ael-demo [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 9 to ael-demo [Feb 22 17:22:18] -- Added extension 's' priority 9 to ael-demo [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 10 to ael-demo [Feb 22 17:22:18] -- Added extension 's' priority 10 to ael-demo [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 11 to ael-demo [Feb 22 17:22:18] -- Added extension 's' priority 11 to ael-demo [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 12 to ael-demo [Feb 22 17:22:18] -- Added extension 's' priority 12 to ael-demo [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '2' priority 1 to ael-demo [Feb 22 17:22:18] -- Added extension '2' priority 1 to ael-demo [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '2' priority 2 to ael-demo [Feb 22 17:22:18] -- Added extension '2' priority 2 to ael-demo [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '3' priority 1 to ael-demo [Feb 22 17:22:18] -- Added extension '3' priority 1 to ael-demo [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '3' priority 2 to ael-demo [Feb 22 17:22:18] -- Added extension '3' priority 2 to ael-demo [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '500' priority 1 to ael-demo [Feb 22 17:22:18] -- Added extension '500' priority 1 to ael-demo [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '500' priority 2 to ael-demo [Feb 22 17:22:18] -- Added extension '500' priority 2 to ael-demo [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '500' priority 3 to ael-demo [Feb 22 17:22:18] -- Added extension '500' priority 3 to ael-demo [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '500' priority 4 to ael-demo [Feb 22 17:22:18] -- Added extension '500' priority 4 to ael-demo [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '600' priority 1 to ael-demo [Feb 22 17:22:18] -- Added extension '600' priority 1 to ael-demo [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '600' priority 2 to ael-demo [Feb 22 17:22:18] -- Added extension '600' priority 2 to ael-demo [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '600' priority 3 to ael-demo [Feb 22 17:22:18] -- Added extension '600' priority 3 to ael-demo [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '600' priority 4 to ael-demo [Feb 22 17:22:18] -- Added extension '600' priority 4 to ael-demo [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_1234' priority 1 to ael-demo [Feb 22 17:22:18] -- Added extension '_1234' priority 1 to ael-demo [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '#' priority 1 to ael-demo [Feb 22 17:22:18] -- Added extension '#' priority 1 to ael-demo [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '#' priority 2 to ael-demo [Feb 22 17:22:18] -- Added extension '#' priority 2 to ael-demo [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 't' priority 1 to ael-demo [Feb 22 17:22:18] -- Added extension 't' priority 1 to ael-demo [Feb 22 17:22:18] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'i' priority 1 to ael-demo [Feb 22 17:22:18] -- Added extension 'i' priority 1 to ael-demo [Feb 22 17:22:18] NOTICE[14736]: pbx_ael.c:3879 pbx_load_module: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'. [Feb 22 17:22:18] DEBUG[14736]: pbx.c:3898 ast_merge_contexts_and_delete: must remove any reg pbx_ael [Feb 22 17:22:18] DEBUG[14736]: pbx.c:5231 __ast_context_destroy: check ctx parkedcalls res_features [Feb 22 17:22:18] NOTICE[14736]: pbx_ael.c:3882 pbx_load_module: AEL load process: merged config file name '/etc/asterisk/extensions.ael'. [Feb 22 17:22:18] NOTICE[14736]: pbx_ael.c:3885 pbx_load_module: AEL load process: verified config file name '/etc/asterisk/extensions.ael'. [Feb 22 17:22:18] pbx_ael.so => (Asterisk Extension Language Compiler) [Feb 22 17:22:18] == Registered file format g723sf, extension(s) g723|g723sf [Feb 22 17:22:18] format_g723.so => (G.723.1 Simple Timestamp File Format) [Feb 22 17:22:18] == Registered file format g729, extension(s) g729 [Feb 22 17:22:18] format_g729.so => (Raw G729 data) [Feb 22 17:22:18] == Parsing '/etc/asterisk/cdr.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/cdr.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] == Registered application 'SoftHangup' [Feb 22 17:22:18] app_softhangup.so => (Hangs up the requested channel) [Feb 22 17:22:18] == Registered application 'ParkAndAnnounce' [Feb 22 17:22:18] app_parkandannounce.so => (Call Parking and Announce Application) [Feb 22 17:22:18] == Parsing '/etc/asterisk/codecs.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/codecs.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] -- codec_zap: using generic PLC [Feb 22 17:22:18] == No hardware transcoders found. [Feb 22 17:22:18] codec_zap.so => (Generic Zaptel Transcoder Codec Translator) [Feb 22 17:22:18] pbx_loopback.so => (Loopback Switch) [Feb 22 17:22:18] == Registered application 'Read' [Feb 22 17:22:18] app_read.so => (Read Variable Application) [Feb 22 17:22:18] == Registered custom function VMCOUNT [Feb 22 17:22:18] == Registered application 'HasVoicemail' [Feb 22 17:22:18] == Registered application 'HasNewVoicemail' [Feb 22 17:22:18] app_hasnewvoicemail.so => (Indicator for whether a voice mailbox has messages in a given folder.) [Feb 22 17:22:18] == Registered application 'PrivacyManager' [Feb 22 17:22:18] app_privacy.so => (Require phone number to be entered, if no CallerID sent) [Feb 22 17:22:18] == Parsing '/etc/asterisk/alarmreceiver.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/alarmreceiver.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] == Registered application 'AlarmReceiver' [Feb 22 17:22:18] app_alarmreceiver.so => (Alarm Receiver for Asterisk) [Feb 22 17:22:18] == Registered file format sln, extension(s) sln|raw [Feb 22 17:22:18] format_sln.so => (Raw Signed Linear Audio support (SLN)) [Feb 22 17:22:18] == Registered custom function CUT [Feb 22 17:22:18] == Registered custom function SORT [Feb 22 17:22:18] func_cut.so => (Cut out information from a string) [Feb 22 17:22:18] == Registered application 'WaitForSilence' [Feb 22 17:22:18] app_waitforsilence.so => (Wait For Silence) [Feb 22 17:22:18] res_convert.so => (File format conversion CLI command) [Feb 22 17:22:18] == Registered custom function MATH [Feb 22 17:22:18] func_math.so => (Mathematical dialplan function) [Feb 22 17:22:18] == Registered application 'SendURL' [Feb 22 17:22:18] app_url.so => (Send URL Applications) [Feb 22 17:22:18] == Parsing '/etc/asterisk/festival.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/festival.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] == Registered application 'Festival' [Feb 22 17:22:18] app_festival.so => (Simple Festival Interface) [Feb 22 17:22:18] == Manager registered action MeetmeMute [Feb 22 17:22:18] == Manager registered action MeetmeUnmute [Feb 22 17:22:18] == Registered application 'MeetMeAdmin' [Feb 22 17:22:18] == Registered application 'MeetMeCount' [Feb 22 17:22:18] == Registered application 'MeetMe' [Feb 22 17:22:18] == Registered application 'SLAStation' [Feb 22 17:22:18] == Registered application 'SLATrunk' [Feb 22 17:22:18] == Parsing '/etc/asterisk/meetme.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/meetme.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] == Parsing '/etc/asterisk/express.demonstration.meetme.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/express.demonstration.meetme.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] app_meetme.so => (MeetMe conference bridge) [Feb 22 17:22:18] == Registered application 'MixMonitor' [Feb 22 17:22:18] == Registered application 'StopMixMonitor' [Feb 22 17:22:18] app_mixmonitor.so => (Mixed Audio Monitoring Application) [Feb 22 17:22:18] == Registered application 'ICES' [Feb 22 17:22:18] app_ices.so => (Encode and Stream via icecast and ices) [Feb 22 17:22:18] == Registered format 'jpg' (JPEG (Joint Picture Experts Group)) [Feb 22 17:22:18] format_jpeg.so => (JPEG (Joint Picture Experts Group) Image Format) [Feb 22 17:22:18] == Registered application 'StackPop' [Feb 22 17:22:18] == Registered application 'Return' [Feb 22 17:22:18] == Registered application 'GosubIf' [Feb 22 17:22:18] == Registered application 'Gosub' [Feb 22 17:22:18] app_stack.so => (Stack Routines) [Feb 22 17:22:18] WARNING[14736]: app_followme.c:297 reload_followme: No follow me config file (followme.conf), so no follow me [Feb 22 17:22:18] == Parsing '/etc/asterisk/dundi.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/dundi.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] DEBUG[14736]: pbx_dundi.c:408 reset_global_eid: Seeding global EID '00:01:03:dc:49:ec' from 'eth0' [Feb 22 17:22:18] == Using TOS bits 0 [Feb 22 17:22:18] == DUNDi Ready and Listening on 0.0.0.0 port 4520 [Feb 22 17:22:18] == Registered custom function DUNDILOOKUP [Feb 22 17:22:18] pbx_dundi.so => (Distributed Universal Number Discovery (DUNDi)) [Feb 22 17:22:18] == Registered application 'MacroExit' [Feb 22 17:22:18] == Registered application 'MacroIf' [Feb 22 17:22:18] == Registered application 'MacroExclusive' [Feb 22 17:22:18] == Registered application 'Macro' [Feb 22 17:22:18] app_macro.so => (Extension Macros) [Feb 22 17:22:18] == Registered application 'RealTimeUpdate' [Feb 22 17:22:18] == Registered application 'RealTime' [Feb 22 17:22:18] app_realtime.so => (Realtime Data Lookup/Rewrite) [Feb 22 17:22:18] == Registered application 'DBdel' [Feb 22 17:22:18] == Registered application 'DBdeltree' [Feb 22 17:22:18] app_db.so => (Database Access Functions) [Feb 22 17:22:18] DEBUG[14736]: channel.c:449 ast_channel_register: Registered handler for 'Feature' (Feature Proxy Channel Driver) [Feb 22 17:22:18] == Registered channel type 'Feature' (Feature Proxy Channel Driver) [Feb 22 17:22:18] chan_features.so => (Feature Proxy Channel) [Feb 22 17:22:18] == Registered application 'Milliwatt' [Feb 22 17:22:18] app_milliwatt.so => (Digital Milliwatt (mu-law) Test Application) [Feb 22 17:22:18] == Parsing '/etc/asterisk/skinny.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/skinny.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] WARNING[14736]: chan_skinny.c:4462 reload_config: Option 'port' at line 5 of skinny.conf has been deprecated. Please use 'bindport' instead. [Feb 22 17:22:18] == Skinny listening on 0.0.0.0:2000 [Feb 22 17:22:18] DEBUG[14736]: channel.c:449 ast_channel_register: Registered handler for 'Skinny' (Skinny Client Control Protocol (Skinny)) [Feb 22 17:22:18] == Registered channel type 'Skinny' (Skinny Client Control Protocol (Skinny)) [Feb 22 17:22:18] == Registered custom function ENUMLOOKUP [Feb 22 17:22:18] == Registered custom function TXTCIDNAME [Feb 22 17:22:18] func_enum.so => (ENUM related dialplan functions) [Feb 22 17:22:18] == Registered application 'DumpChan' [Feb 22 17:22:18] app_dumpchan.so => (Dump Info About The Calling Channel) [Feb 22 17:22:18] == Parsing '/etc/asterisk/phone.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/phone.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] DEBUG[14736]: channel.c:449 ast_channel_register: Registered handler for 'Phone' (Standard Linux Telephony API Driver) [Feb 22 17:22:18] == Registered channel type 'Phone' (Standard Linux Telephony API Driver) [Feb 22 17:22:18] chan_phone.so => (Linux Telephony API Support) [Feb 22 17:22:18] == Registered application 'TestClient' [Feb 22 17:22:18] == Registered application 'TestServer' [Feb 22 17:22:18] app_test.so => (Interface Test Application) [Feb 22 17:22:18] == Registered custom function CURL [Feb 22 17:22:18] func_curl.so => (Load external URL) [Feb 22 17:22:18] == Registered application 'Morsecode' [Feb 22 17:22:18] app_morsecode.so => (Morse code) [Feb 22 17:22:18] == Registered application 'Log' [Feb 22 17:22:18] == Registered application 'Verbose' [Feb 22 17:22:18] app_verbose.so => (Send verbose output) [Feb 22 17:22:18] == Parsing '/etc/asterisk/cdr_manager.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/cdr_manager.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] == Registered application 'SendText' [Feb 22 17:22:18] app_sendtext.so => (Send Text Applications) [Feb 22 17:22:18] == Registered custom function ISNULL [Feb 22 17:22:18] == Registered custom function SET [Feb 22 17:22:18] == Registered custom function EXISTS [Feb 22 17:22:18] == Registered custom function IF [Feb 22 17:22:18] == Registered custom function IFTIME [Feb 22 17:22:18] func_logic.so => (Logical dialplan functions) [Feb 22 17:22:18] == Parsing '/etc/asterisk/codecs.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/codecs.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] -- codec_lpc10: using generic PLC [Feb 22 17:22:18] WARNING[14736]: translate.c:675 __ast_register_translator: plc_samples 180 format 6 [Feb 22 17:22:18] == Registered translator 'lpc10tolin' from format lpc10 to slin, cost 2 [Feb 22 17:22:18] DEBUG[14736]: translate.c:425 rebuild_matrix: Resetting translation matrix [Feb 22 17:22:18] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Feb 22 17:22:18] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:18] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 6 [Feb 22 17:22:18] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:18] == Registered translator 'lintolpc10' from format slin to lpc10, cost 3 [Feb 22 17:22:18] DEBUG[14736]: translate.c:425 rebuild_matrix: Resetting translation matrix [Feb 22 17:22:18] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Feb 22 17:22:18] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Feb 22 17:22:18] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:18] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 6 [Feb 22 17:22:18] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:18] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6 [Feb 22 17:22:18] codec_lpc10.so => (LPC10 2.4kbps Coder/Decoder) [Feb 22 17:22:18] == Registered custom function CDR [Feb 22 17:22:18] func_cdr.so => (CDR dialplan function) [Feb 22 17:22:18] == Registered custom function RAND [Feb 22 17:22:18] func_rand.so => (Random number dialplan function) [Feb 22 17:22:18] == Registered application 'Echo' [Feb 22 17:22:18] app_echo.so => (Simple Echo Application) [Feb 22 17:22:18] == Registered application 'DeadAGI' [Feb 22 17:22:18] == Registered application 'EAGI' [Feb 22 17:22:18] == Registered application 'AGI' [Feb 22 17:22:18] res_agi.so => (Asterisk Gateway Interface (AGI)) [Feb 22 17:22:18] == Registered application 'Record' [Feb 22 17:22:18] app_record.so => (Trivial Record Application) [Feb 22 17:22:18] == Registered application 'ZapBarge' [Feb 22 17:22:18] app_zapbarge.so => (Barge in on Zap channel application) [Feb 22 17:22:18] == Parsing '/etc/asterisk/mgcp.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/mgcp.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] == MGCP Listening on 0.0.0.0:2727 [Feb 22 17:22:18] == Using TOS bits 0 [Feb 22 17:22:18] DEBUG[14736]: channel.c:449 ast_channel_register: Registered handler for 'MGCP' (Media Gateway Control Protocol (MGCP)) [Feb 22 17:22:18] == Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP)) [Feb 22 17:22:18] chan_mgcp.so => (Media Gateway Control Protocol (MGCP)) [Feb 22 17:22:18] == Parsing '/etc/asterisk/codecs.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/codecs.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] -- codec_gsm: using generic PLC [Feb 22 17:22:18] WARNING[14736]: translate.c:675 __ast_register_translator: plc_samples 160 format 6 [Feb 22 17:22:18] == Registered translator 'gsmtolin' from format gsm to slin, cost 1 [Feb 22 17:22:18] DEBUG[14736]: translate.c:425 rebuild_matrix: Resetting translation matrix [Feb 22 17:22:18] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Feb 22 17:22:18] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6 [Feb 22 17:22:18] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from g723 to unknown, via 6 [Feb 22 17:22:18] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Feb 22 17:22:18] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Feb 22 17:22:18] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:18] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 6 [Feb 22 17:22:18] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:18] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6 [Feb 22 17:22:18] == Registered translator 'lintogsm' from format slin to gsm, cost 3 [Feb 22 17:22:18] DEBUG[14736]: translate.c:425 rebuild_matrix: Resetting translation matrix [Feb 22 17:22:18] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Feb 22 17:22:18] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6 [Feb 22 17:22:18] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from g723 to unknown, via 6 [Feb 22 17:22:18] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Feb 22 17:22:18] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Feb 22 17:22:18] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Feb 22 17:22:18] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 6 [Feb 22 17:22:18] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:18] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 6 [Feb 22 17:22:18] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 6 [Feb 22 17:22:18] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:18] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6 [Feb 22 17:22:18] codec_gsm.so => (GSM Coder/Decoder) [Feb 22 17:22:18] == Registered custom function IAXPEER [Feb 22 17:22:18] WARNING[14736]: cli.c:1434 __ast_cli_register: Command 'iax2 show cache' already registered (or something close enough) [Feb 22 17:22:18] WARNING[14736]: cli.c:1434 __ast_cli_register: Command 'iax2 show channels' already registered (or something close enough) [Feb 22 17:22:18] WARNING[14736]: cli.c:1434 __ast_cli_register: Command 'iax2 show firmware' already registered (or something close enough) [Feb 22 17:22:18] WARNING[14736]: cli.c:1434 __ast_cli_register: Command 'iax2 show netstats' already registered (or something close enough) [Feb 22 17:22:18] WARNING[14736]: cli.c:1434 __ast_cli_register: Command 'iax2 show peers' already registered (or something close enough) [Feb 22 17:22:18] WARNING[14736]: cli.c:1434 __ast_cli_register: Command 'iax2 show registry' already registered (or something close enough) [Feb 22 17:22:18] WARNING[14736]: cli.c:1434 __ast_cli_register: Command 'iax2 show stats' already registered (or something close enough) [Feb 22 17:22:18] WARNING[14736]: cli.c:1434 __ast_cli_register: Command 'iax2 show threads' already registered (or something close enough) [Feb 22 17:22:18] WARNING[14736]: cli.c:1434 __ast_cli_register: Command 'iax2 show users' already registered (or something close enough) [Feb 22 17:22:18] == Registered application 'IAX2Provision' [Feb 22 17:22:18] == Manager registered action IAXpeers [Feb 22 17:22:18] == Manager registered action IAXnetstats [Feb 22 17:22:18] == Parsing '/etc/asterisk/iax.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/iax.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] == Parsing '/etc/asterisk/express.demonstration.extensions.iax.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.iax.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] == Parsing '/etc/asterisk/express.demonstration.extensions.iax.605.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.iax.605.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] == Parsing '/etc/asterisk/express.demonstration.extensions.iax.606.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.iax.606.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] == Parsing '/etc/asterisk/express.demonstration.extensions.iax.614.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.iax.614.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] == Parsing '/etc/asterisk/express.demonstration.extensions.iax.616.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.iax.616.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] == Parsing '/etc/asterisk/express.demonstration.extensions.iax.651.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.iax.651.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] == Parsing '/etc/asterisk/express.demonstration.extensions.iax.698.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.iax.698.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] WARNING[14736]: acl.c:310 ast_str2tos: TOS value lowdelay is deprecated. Please see doc/ip-tos.txt for more information. [Feb 22 17:22:18] == Using TOS bits 16 [Feb 22 17:22:18] == Binding IAX2 to '24.123.23.170:4569' [Feb 22 17:22:18] WARNING[14736]: acl.c:310 ast_str2tos: TOS value lowdelay is deprecated. Please see doc/ip-tos.txt for more information. [Feb 22 17:22:18] > doing dnsmgr_lookup for '192.168.1.10' [Feb 22 17:22:18] > doing dnsmgr_lookup for '192.168.1.10' [Feb 22 17:22:18] DEBUG[14736]: db.c:197 ast_db_get: Unable to find key 'dell_8200_to_unifiedpaging' in family 'IAX/Registry' [Feb 22 17:22:18] > doing dnsmgr_lookup for 'switch-1.nufone.net' [Feb 22 17:22:18] > doing dnsmgr_lookup for '192.168.1.170' [Feb 22 17:22:18] > doing dnsmgr_lookup for '192.168.1.159' [Feb 22 17:22:18] DEBUG[14736]: db.c:197 ast_db_get: Unable to find key '605' in family 'IAX/Registry' [Feb 22 17:22:18] -- Seeding '606' at 192.168.1.166:4569 for 60 [Feb 22 17:22:18] DEBUG[14736]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel IAX2/606 [Feb 22 17:22:18] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for IAX2 - 606 [Feb 22 17:22:18] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for IAX2/606 - state 4 (Invalid) [Feb 22 17:22:18] -- Seeding '614' at 192.168.1.167:4569 for 60 [Feb 22 17:22:18] DEBUG[14736]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel IAX2/614 [Feb 22 17:22:18] -- Seeding '616' at 74.133.61.143:4569 for 60 [Feb 22 17:22:18] DEBUG[14736]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel IAX2/616 [Feb 22 17:22:18] -- Seeding '651' at 192.168.1.169:4569 for 60 [Feb 22 17:22:18] DEBUG[14751]: app_queue.c:546 changethread: Device 'IAX2/606' changed to state '4' (Invalid) but we don't care because they're not a member of any queue. [Feb 22 17:22:18] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for IAX2 - 614 [Feb 22 17:22:18] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for IAX2/614 - state 4 (Invalid) [Feb 22 17:22:18] DEBUG[14736]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel IAX2/651 [Feb 22 17:22:18] DEBUG[14736]: db.c:197 ast_db_get: Unable to find key '698' in family 'IAX/Registry' [Feb 22 17:22:18] DEBUG[14736]: channel.c:449 ast_channel_register: Registered handler for 'IAX2' (Inter Asterisk eXchange Driver (Ver 2)) [Feb 22 17:22:18] == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2)) [Feb 22 17:22:18] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for IAX2 - 616 [Feb 22 17:22:18] DEBUG[14741]: chan_iax2.c:9600 iax2_devicestate: Checking device state for device 616 [Feb 22 17:22:18] DEBUG[14741]: chan_iax2.c:9608 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 616? addr=-1891793590, defaddr=0 maxms=0, lastms=0 [Feb 22 17:22:18] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for IAX2/616 - state 1 (Not in use) [Feb 22 17:22:18] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for IAX2 - 651 [Feb 22 17:22:18] DEBUG[14741]: chan_iax2.c:9600 iax2_devicestate: Checking device state for device 651 [Feb 22 17:22:18] DEBUG[14741]: chan_iax2.c:9608 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 651? addr=-1459509056, defaddr=0 maxms=0, lastms=0 [Feb 22 17:22:18] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for IAX2/651 - state 1 (Not in use) [Feb 22 17:22:18] DEBUG[14752]: app_queue.c:546 changethread: Device 'IAX2/614' changed to state '4' (Invalid) but we don't care because they're not a member of any queue. [Feb 22 17:22:18] DEBUG[14756]: app_queue.c:546 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. [Feb 22 17:22:18] DEBUG[14762]: app_queue.c:546 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. [Feb 22 17:22:18] == 10 helper threaads started [Feb 22 17:22:18] == IAX Ready and Listening [Feb 22 17:22:18] == Loaded firmware 'iaxy.bin' [Feb 22 17:22:18] == Parsing '/etc/asterisk/iaxprov.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/iaxprov.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] WARNING[14736]: acl.c:310 ast_str2tos: TOS value lowdelay is deprecated. Please see doc/ip-tos.txt for more information. [Feb 22 17:22:18] -- Loaded provisioning template 'default' [Feb 22 17:22:18] chan_iax2.so => (Inter Asterisk eXchange (Ver 2)) [Feb 22 17:22:18] == Registered application 'Dial' [Feb 22 17:22:18] == Registered application 'RetryDial' [Feb 22 17:22:18] app_dial.so => (Dialing Application) [Feb 22 17:22:18] == Registered application 'ADSIProg' [Feb 22 17:22:18] app_adsiprog.so => (Asterisk ADSI Programming Application) [Feb 22 17:22:18] == Registered custom function REALTIME [Feb 22 17:22:18] func_realtime.so => (Read/Write values from a RealTime repository) [Feb 22 17:22:18] == Registered application 'Exec' [Feb 22 17:22:18] == Registered application 'TryExec' [Feb 22 17:22:18] == Registered application 'ExecIf' [Feb 22 17:22:18] app_exec.so => (Executes dialplan applications) [Feb 22 17:22:18] == Registered file format h264, extension(s) h264 [Feb 22 17:22:18] format_h264.so => (Raw H.264 data) [Feb 22 17:22:18] == Parsing '/etc/asterisk/sip.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/sip.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] == Parsing '/etc/asterisk/express.demonstration.extensions.sip.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] == Parsing '/etc/asterisk/express.demonstration.extensions.sip.520.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.520.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] == Parsing '/etc/asterisk/express.demonstration.extensions.sip.521.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.521.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] == Parsing '/etc/asterisk/express.demonstration.extensions.sip.522.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.522.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] == Parsing '/etc/asterisk/express.demonstration.extensions.sip.523.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.523.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] == Parsing '/etc/asterisk/express.demonstration.extensions.sip.524.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.524.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] == Parsing '/etc/asterisk/express.demonstration.extensions.sip.525.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.525.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] == Parsing '/etc/asterisk/express.demonstration.extensions.sip.526.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.526.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] == Parsing '/etc/asterisk/express.demonstration.extensions.sip.528.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.528.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] == Parsing '/etc/asterisk/express.demonstration.extensions.sip.529.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.529.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] == Parsing '/etc/asterisk/express.demonstration.extensions.sip.540.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.540.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] == Parsing '/etc/asterisk/express.demonstration.extensions.sip.541.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.541.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] == Parsing '/etc/asterisk/express.demonstration.extensions.sip.550.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.550.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] == Parsing '/etc/asterisk/express.demonstration.extensions.sip.551.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.551.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] == Parsing '/etc/asterisk/express.demonstration.extensions.sip.592.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.592.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] == Parsing '/etc/asterisk/express.demonstration.extensions.sip.593.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.593.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] == Parsing '/etc/asterisk/express.demonstration.extensions.sip.594.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.594.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] == Parsing '/etc/asterisk/express.demonstration.extensions.sip.595.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.595.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] == Parsing '/etc/asterisk/express.demonstration.extensions.sip.596.conf': [Feb 22 17:22:18] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.596.conf [Feb 22 17:22:18] Found [Feb 22 17:22:18] WARNING[14736]: chan_sip.c:15252 handle_common_options: insecure=very at line 315 is deprecated; use insecure=port,invite instead [Feb 22 17:22:18] WARNING[14736]: acl.c:182 ast_append_ha: sip.broadvoice.com is not a valid IP [Feb 22 17:22:19] WARNING[14736]: acl.c:182 ast_append_ha: sip.broadvoice.com is not a valid IP [Feb 22 17:22:19] -- SIP Seeding peer from astdb: '510' at 510@192.168.1.165:5060 for 60 [Feb 22 17:22:19] -- SIP Seeding peer from astdb: '511' at 511@192.168.1.165:5060 for 60 [Feb 22 17:22:19] -- SIP Seeding peer from astdb: '512' at 512@192.168.1.165:5060 for 60 [Feb 22 17:22:19] -- SIP Seeding peer from astdb: '513' at 513@192.168.1.165:5060 for 60 [Feb 22 17:22:19] -- SIP Seeding peer from astdb: '514' at 514@192.168.1.165:5060 for 60 [Feb 22 17:22:19] -- SIP Seeding peer from astdb: '515' at 515@192.168.1.165:5060 for 60 [Feb 22 17:22:19] DEBUG[14736]: db.c:197 ast_db_get: Unable to find key '520' in family 'SIP/Registry' [Feb 22 17:22:19] DEBUG[14736]: db.c:197 ast_db_get: Unable to find key '530' in family 'SIP/Registry' [Feb 22 17:22:19] DEBUG[14736]: db.c:197 ast_db_get: Unable to find key '531' in family 'SIP/Registry' [Feb 22 17:22:19] DEBUG[14736]: db.c:197 ast_db_get: Unable to find key '532' in family 'SIP/Registry' [Feb 22 17:22:19] DEBUG[14736]: db.c:197 ast_db_get: Unable to find key '534' in family 'SIP/Registry' [Feb 22 17:22:19] DEBUG[14736]: db.c:197 ast_db_get: Unable to find key '535' in family 'SIP/Registry' [Feb 22 17:22:19] DEBUG[14736]: db.c:197 ast_db_get: Unable to find key '536' in family 'SIP/Registry' [Feb 22 17:22:19] DEBUG[14736]: db.c:197 ast_db_get: Unable to find key '537' in family 'SIP/Registry' [Feb 22 17:22:19] -- SIP Seeding peer from astdb: '540' at 540@192.168.1.62:5060 for 3600 [Feb 22 17:22:19] DEBUG[14736]: db.c:197 ast_db_get: Unable to find key '597' in family 'SIP/Registry' [Feb 22 17:22:19] DEBUG[14736]: db.c:197 ast_db_get: Unable to find key '598' in family 'SIP/Registry' [Feb 22 17:22:19] DEBUG[14736]: db.c:197 ast_db_get: Unable to find key '599' in family 'SIP/Registry' [Feb 22 17:22:19] DEBUG[14736]: db.c:197 ast_db_get: Unable to find key '1001' in family 'SIP/Registry' [Feb 22 17:22:19] DEBUG[14736]: db.c:197 ast_db_get: Unable to find key '1002' in family 'SIP/Registry' [Feb 22 17:22:19] DEBUG[14736]: db.c:197 ast_db_get: Unable to find key '1003' in family 'SIP/Registry' [Feb 22 17:22:19] DEBUG[14736]: db.c:197 ast_db_get: Unable to find key '1004' in family 'SIP/Registry' [Feb 22 17:22:19] DEBUG[14736]: db.c:197 ast_db_get: Unable to find key '521' in family 'SIP/Registry' [Feb 22 17:22:19] -- SIP Seeding peer from astdb: '522' at 522@192.168.1.95:5060 for 60 [Feb 22 17:22:19] -- SIP Seeding peer from astdb: '523' at 523@192.168.1.85:5060 for 60 [Feb 22 17:22:19] DEBUG[14736]: db.c:197 ast_db_get: Unable to find key '524' in family 'SIP/Registry' [Feb 22 17:22:19] -- SIP Seeding peer from astdb: '525' at 525@192.168.1.76:5060 for 60 [Feb 22 17:22:19] -- SIP Seeding peer from astdb: '526' at 526@192.168.1.66:5060 for 60 [Feb 22 17:22:19] -- SIP Seeding peer from astdb: '528' at 528@192.168.1.93:5060 for 60 [Feb 22 17:22:19] -- SIP Seeding peer from astdb: '529' at 529@192.168.1.98:5060 for 60 [Feb 22 17:22:19] -- SIP Seeding peer from astdb: '541' at 541@192.168.1.90:5060 for 3600 [Feb 22 17:22:19] DEBUG[14736]: db.c:197 ast_db_get: Unable to find key '550' in family 'SIP/Registry' [Feb 22 17:22:19] DEBUG[14736]: db.c:197 ast_db_get: Unable to find key '551' in family 'SIP/Registry' [Feb 22 17:22:19] -- SIP Seeding peer from astdb: '592' at 592@68.58.36.157:5060 for 3600 [Feb 22 17:22:19] DEBUG[14736]: db.c:197 ast_db_get: Unable to find key '593' in family 'SIP/Registry' [Feb 22 17:22:19] DEBUG[14736]: db.c:197 ast_db_get: Unable to find key '594' in family 'SIP/Registry' [Feb 22 17:22:19] -- SIP Seeding peer from astdb: '595' at 595@192.168.1.86:5060 for 3600 [Feb 22 17:22:19] -- SIP Seeding peer from astdb: '596' at 596@192.168.1.86:5062 for 3600 [Feb 22 17:22:19] == SIP Listening on 24.123.23.170:5060 [Feb 22 17:22:19] == Using SIP TOS: none [Feb 22 17:22:19] == Parsing '/etc/asterisk/sip_notify.conf': [Feb 22 17:22:19] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/sip_notify.conf [Feb 22 17:22:19] Found [Feb 22 17:22:19] DEBUG[14736]: channel.c:449 ast_channel_register: Registered handler for 'SIP' (Session Initiation Protocol (SIP)) [Feb 22 17:22:19] == Registered channel type 'SIP' (Session Initiation Protocol (SIP)) [Feb 22 17:22:19] == Registered application 'SIPDtmfMode' [Feb 22 17:22:19] == Registered application 'SIPAddHeader' [Feb 22 17:22:19] == Registered custom function SIP_HEADER [Feb 22 17:22:19] == Registered custom function SIPPEER [Feb 22 17:22:19] == Registered custom function SIPCHANINFO [Feb 22 17:22:19] == Registered custom function CHECKSIPDOMAIN [Feb 22 17:22:19] == Manager registered action SIPpeers [Feb 22 17:22:19] == Manager registered action SIPshowpeer [Feb 22 17:22:19] chan_sip.so => (Session Initiation Protocol (SIP)) [Feb 22 17:22:19] == Parsing '/etc/asterisk/extensions.conf': [Feb 22 17:22:19] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/extensions.conf [Feb 22 17:22:19] Found [Feb 22 17:22:19] == Parsing '/etc/asterisk/express.demonstration.extensions.meetme.conf': [Feb 22 17:22:19] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.meetme.conf [Feb 22 17:22:19] Found [Feb 22 17:22:19] == Parsing '/etc/asterisk/express.demonstration.dnis.conf': [Feb 22 17:22:19] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/express.demonstration.dnis.conf [Feb 22 17:22:19] Found [Feb 22 17:22:19] == Parsing '/etc/asterisk/express.demonstration.extensions.meetme.conf': [Feb 22 17:22:19] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.meetme.conf [Feb 22 17:22:19] Found [Feb 22 17:22:19] == Setting global variable 'CONSOLE' to 'Console/dsp' [Feb 22 17:22:19] == Setting global variable 'IAXINFO' to 'guest' [Feb 22 17:22:19] == Setting global variable 'TRUNK' to 'Zap/g2' [Feb 22 17:22:19] == Setting global variable 'TRUNKMSD' to '1' [Feb 22 17:22:19] == Setting global variable 'DIAL_TIMEOUT' to '20' [Feb 22 17:22:19] == Setting global variable 'SMVOICE_DIAL_TIMEOUT' to '60' [Feb 22 17:22:19] == Setting global variable 'SMVOICE_ANNOUNCE_CALLER' to '1' [Feb 22 17:22:19] == Setting global variable 'SMVOICE_DIAL_LONG_TIMEOUT' to '120' [Feb 22 17:22:19] == Setting global variable 'OPERATOR' to '510' [Feb 22 17:22:19] == Setting global variable 'OPERATOR_TECHNOLOGY' to 'SIP' [Feb 22 17:22:19] == Setting global variable 'SUPPORT' to '216' [Feb 22 17:22:19] == Setting global variable 'SALES' to '217' [Feb 22 17:22:19] == Setting global variable 'INTERCOM' to 'Zap/8' [Feb 22 17:22:19] == Setting global variable 'SMVOICE_ONHOLD' to '' [Feb 22 17:22:19] DEBUG[14736]: pbx.c:3824 __ast_context_create: Registered context 'iaxtel700' [Feb 22 17:22:19] -- Registered extension context 'iaxtel700' [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_91700XXXXXXX' priority 1 to iaxtel700 [Feb 22 17:22:19] -- Added extension '_91700XXXXXXX' priority 1 to iaxtel700 [Feb 22 17:22:19] DEBUG[14736]: pbx.c:3824 __ast_context_create: Registered context 'iaxprovider' [Feb 22 17:22:19] -- Registered extension context 'iaxprovider' [Feb 22 17:22:19] DEBUG[14736]: pbx.c:3824 __ast_context_create: Registered context 'trunkint' [Feb 22 17:22:19] -- Registered extension context 'trunkint' [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_9011.' priority 1 to trunkint [Feb 22 17:22:19] -- Added extension '_9011.' priority 1 to trunkint [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_9011.' priority 2 to trunkint [Feb 22 17:22:19] -- Added extension '_9011.' priority 2 to trunkint [Feb 22 17:22:19] DEBUG[14736]: pbx.c:3824 __ast_context_create: Registered context 'trunkld' [Feb 22 17:22:19] -- Registered extension context 'trunkld' [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_91NXXNXXXXXX' priority 1 to trunkld [Feb 22 17:22:19] -- Added extension '_91NXXNXXXXXX' priority 1 to trunkld [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_91NXXNXXXXXX' priority 2 to trunkld [Feb 22 17:22:19] -- Added extension '_91NXXNXXXXXX' priority 2 to trunkld [Feb 22 17:22:19] DEBUG[14736]: pbx.c:3824 __ast_context_create: Registered context 'trunklocal' [Feb 22 17:22:19] -- Registered extension context 'trunklocal' [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_9NXXXXXX' priority 1 to trunklocal [Feb 22 17:22:19] -- Added extension '_9NXXXXXX' priority 1 to trunklocal [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_9NXXXXXX' priority 2 to trunklocal [Feb 22 17:22:19] -- Added extension '_9NXXXXXX' priority 2 to trunklocal [Feb 22 17:22:19] DEBUG[14736]: pbx.c:3824 __ast_context_create: Registered context 'trunktollfree' [Feb 22 17:22:19] -- Registered extension context 'trunktollfree' [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_91800NXXXXXX' priority 1 to trunktollfree [Feb 22 17:22:19] -- Added extension '_91800NXXXXXX' priority 1 to trunktollfree [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_91800NXXXXXX' priority 2 to trunktollfree [Feb 22 17:22:19] -- Added extension '_91800NXXXXXX' priority 2 to trunktollfree [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_91888NXXXXXX' priority 1 to trunktollfree [Feb 22 17:22:19] -- Added extension '_91888NXXXXXX' priority 1 to trunktollfree [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_91888NXXXXXX' priority 2 to trunktollfree [Feb 22 17:22:19] -- Added extension '_91888NXXXXXX' priority 2 to trunktollfree [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_91877NXXXXXX' priority 1 to trunktollfree [Feb 22 17:22:19] -- Added extension '_91877NXXXXXX' priority 1 to trunktollfree [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_91877NXXXXXX' priority 2 to trunktollfree [Feb 22 17:22:19] -- Added extension '_91877NXXXXXX' priority 2 to trunktollfree [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_91866NXXXXXX' priority 1 to trunktollfree [Feb 22 17:22:19] -- Added extension '_91866NXXXXXX' priority 1 to trunktollfree [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_91866NXXXXXX' priority 2 to trunktollfree [Feb 22 17:22:19] -- Added extension '_91866NXXXXXX' priority 2 to trunktollfree [Feb 22 17:22:19] DEBUG[14736]: pbx.c:3824 __ast_context_create: Registered context 'international' [Feb 22 17:22:19] -- Registered extension context 'international' [Feb 22 17:22:19] -- Including context 'longdistance' in context 'international' [Feb 22 17:22:19] -- Including context 'trunkint' in context 'international' [Feb 22 17:22:19] DEBUG[14736]: pbx.c:3824 __ast_context_create: Registered context 'longdistance' [Feb 22 17:22:19] -- Registered extension context 'longdistance' [Feb 22 17:22:19] -- Including context 'local' in context 'longdistance' [Feb 22 17:22:19] -- Including context 'trunkld' in context 'longdistance' [Feb 22 17:22:19] DEBUG[14736]: pbx.c:3824 __ast_context_create: Registered context 'local' [Feb 22 17:22:19] -- Registered extension context 'local' [Feb 22 17:22:19] -- Including context 'default' in context 'local' [Feb 22 17:22:19] -- Including context 'parkedcalls' in context 'local' [Feb 22 17:22:19] -- Including context 'trunklocal' in context 'local' [Feb 22 17:22:19] -- Including context 'iaxtel700' in context 'local' [Feb 22 17:22:19] -- Including context 'trunktollfree' in context 'local' [Feb 22 17:22:19] -- Including context 'iaxprovider' in context 'local' [Feb 22 17:22:19] DEBUG[14736]: pbx.c:3824 __ast_context_create: Registered context 'macro-stdexten' [Feb 22 17:22:19] -- Registered extension context 'macro-stdexten' [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 1 to macro-stdexten [Feb 22 17:22:19] -- Added extension 's' priority 1 to macro-stdexten [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 2 to macro-stdexten [Feb 22 17:22:19] -- Added extension 's' priority 2 to macro-stdexten [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's-NOANSWER' priority 1 to macro-stdexten [Feb 22 17:22:19] -- Added extension 's-NOANSWER' priority 1 to macro-stdexten [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's-NOANSWER' priority 2 to macro-stdexten [Feb 22 17:22:19] -- Added extension 's-NOANSWER' priority 2 to macro-stdexten [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's-BUSY' priority 1 to macro-stdexten [Feb 22 17:22:19] -- Added extension 's-BUSY' priority 1 to macro-stdexten [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's-BUSY' priority 2 to macro-stdexten [Feb 22 17:22:19] -- Added extension 's-BUSY' priority 2 to macro-stdexten [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_s-.' priority 1 to macro-stdexten [Feb 22 17:22:19] -- Added extension '_s-.' priority 1 to macro-stdexten [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'a' priority 1 to macro-stdexten [Feb 22 17:22:19] -- Added extension 'a' priority 1 to macro-stdexten [Feb 22 17:22:19] DEBUG[14736]: pbx.c:3824 __ast_context_create: Registered context 'demo' [Feb 22 17:22:19] -- Registered extension context 'demo' [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 1 to demo [Feb 22 17:22:19] -- Added extension 's' priority 1 to demo [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 2 to demo [Feb 22 17:22:19] -- Added extension 's' priority 2 to demo [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 3 to demo [Feb 22 17:22:19] -- Added extension 's' priority 3 to demo [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 4 to demo [Feb 22 17:22:19] -- Added extension 's' priority 4 to demo [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 5 to demo [Feb 22 17:22:19] -- Added extension 's' priority 5 to demo [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 6 to demo [Feb 22 17:22:19] -- Added extension 's' priority 6 to demo [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '2' priority 1 to demo [Feb 22 17:22:19] -- Added extension '2' priority 1 to demo [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '2' priority 2 to demo [Feb 22 17:22:19] -- Added extension '2' priority 2 to demo [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '3' priority 1 to demo [Feb 22 17:22:19] -- Added extension '3' priority 1 to demo [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '3' priority 2 to demo [Feb 22 17:22:19] -- Added extension '3' priority 2 to demo [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '1000' priority 1 to demo [Feb 22 17:22:19] -- Added extension '1000' priority 1 to demo [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '1234' priority 1 to demo [Feb 22 17:22:19] -- Added extension '1234' priority 1 to demo [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '1234' priority 2 to demo [Feb 22 17:22:19] -- Added extension '1234' priority 2 to demo [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '1235' priority 1 to demo [Feb 22 17:22:19] -- Added extension '1235' priority 1 to demo [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '1236' priority 1 to demo [Feb 22 17:22:19] -- Added extension '1236' priority 1 to demo [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '1236' priority 2 to demo [Feb 22 17:22:19] -- Added extension '1236' priority 2 to demo [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '#' priority 1 to demo [Feb 22 17:22:19] -- Added extension '#' priority 1 to demo [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '#' priority 2 to demo [Feb 22 17:22:19] -- Added extension '#' priority 2 to demo [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 't' priority 1 to demo [Feb 22 17:22:19] -- Added extension 't' priority 1 to demo [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'i' priority 1 to demo [Feb 22 17:22:19] -- Added extension 'i' priority 1 to demo [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '500' priority 1 to demo [Feb 22 17:22:19] -- Added extension '500' priority 1 to demo [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '500' priority 2 to demo [Feb 22 17:22:19] -- Added extension '500' priority 2 to demo [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '500' priority 3 to demo [Feb 22 17:22:19] -- Added extension '500' priority 3 to demo [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '500' priority 4 to demo [Feb 22 17:22:19] -- Added extension '500' priority 4 to demo [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '600' priority 1 to demo [Feb 22 17:22:19] -- Added extension '600' priority 1 to demo [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '600' priority 2 to demo [Feb 22 17:22:19] -- Added extension '600' priority 2 to demo [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '600' priority 3 to demo [Feb 22 17:22:19] -- Added extension '600' priority 3 to demo [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '600' priority 4 to demo [Feb 22 17:22:19] -- Added extension '600' priority 4 to demo [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '8500' priority 1 to demo [Feb 22 17:22:19] -- Added extension '8500' priority 1 to demo [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '8500' priority 2 to demo [Feb 22 17:22:19] -- Added extension '8500' priority 2 to demo [Feb 22 17:22:19] DEBUG[14736]: pbx.c:3824 __ast_context_create: Registered context 'default' [Feb 22 17:22:19] -- Registered extension context 'default' [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 1 to default [Feb 22 17:22:19] -- Added extension 's' priority 1 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 2 to default [Feb 22 17:22:19] -- Added extension 's' priority 2 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 3 to default [Feb 22 17:22:19] -- Added extension 's' priority 3 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 4 to default [Feb 22 17:22:19] -- Added extension 's' priority 4 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 5 to default [Feb 22 17:22:19] -- Added extension 's' priority 5 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 6 to default [Feb 22 17:22:19] -- Added extension 's' priority 6 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 7 to default [Feb 22 17:22:19] -- Added extension 's' priority 7 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 8 to default [Feb 22 17:22:19] -- Added extension 's' priority 8 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 9 to default [Feb 22 17:22:19] -- Added extension 's' priority 9 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '3173241051' priority 1 to default [Feb 22 17:22:19] -- Added extension '3173241051' priority 1 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '3173241052' priority 1 to default [Feb 22 17:22:19] -- Added extension '3173241052' priority 1 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'operator' priority 1 to default [Feb 22 17:22:19] -- Added extension 'operator' priority 1 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'operator' priority 2 to default [Feb 22 17:22:19] -- Added extension 'operator' priority 2 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'operator' priority 3 to default [Feb 22 17:22:19] -- Added extension 'operator' priority 3 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'operator' priority 4 to default [Feb 22 17:22:19] -- Added extension 'operator' priority 4 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'operator' priority 5 to default [Feb 22 17:22:19] -- Added extension 'operator' priority 5 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'operator' priority 6 to default [Feb 22 17:22:19] -- Added extension 'operator' priority 6 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'operator-CHANUNAVAIL' priority 1 to default [Feb 22 17:22:19] -- Added extension 'operator-CHANUNAVAIL' priority 1 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'operator-CONGESTION' priority 1 to default [Feb 22 17:22:19] -- Added extension 'operator-CONGESTION' priority 1 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'operator-NOANSWER' priority 1 to default [Feb 22 17:22:19] -- Added extension 'operator-NOANSWER' priority 1 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'operator-BUSY' priority 1 to default [Feb 22 17:22:19] -- Added extension 'operator-BUSY' priority 1 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'operator' priority 102 to default [Feb 22 17:22:19] -- Added extension 'operator' priority 102 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '0' priority 1 to default [Feb 22 17:22:19] -- Added extension '0' priority 1 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '0' priority 2 to default [Feb 22 17:22:19] -- Added extension '0' priority 2 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '0' priority 3 to default [Feb 22 17:22:19] -- Added extension '0' priority 3 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '1' priority 1 to default [Feb 22 17:22:19] -- Added extension '1' priority 1 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '1' priority 2 to default [Feb 22 17:22:19] -- Added extension '1' priority 2 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '1' priority 3 to default [Feb 22 17:22:19] -- Added extension '1' priority 3 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '1' priority 4 to default [Feb 22 17:22:19] -- Added extension '1' priority 4 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '2' priority 1 to default [Feb 22 17:22:19] -- Added extension '2' priority 1 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '2' priority 2 to default [Feb 22 17:22:19] -- Added extension '2' priority 2 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '2' priority 3 to default [Feb 22 17:22:19] -- Added extension '2' priority 3 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '2' priority 4 to default [Feb 22 17:22:19] -- Added extension '2' priority 4 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '3' priority 1 to default [Feb 22 17:22:19] -- Added extension '3' priority 1 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '3' priority 2 to default [Feb 22 17:22:19] -- Added extension '3' priority 2 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '3' priority 3 to default [Feb 22 17:22:19] -- Added extension '3' priority 3 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '4' priority 1 to default [Feb 22 17:22:19] -- Added extension '4' priority 1 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '4' priority 2 to default [Feb 22 17:22:19] -- Added extension '4' priority 2 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '4' priority 3 to default [Feb 22 17:22:19] -- Added extension '4' priority 3 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '4' priority 4 to default [Feb 22 17:22:19] -- Added extension '4' priority 4 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '22' priority 1 to default [Feb 22 17:22:19] -- Added extension '22' priority 1 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '22' priority 2 to default [Feb 22 17:22:19] -- Added extension '22' priority 2 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '711' priority 1 to default [Feb 22 17:22:19] -- Added extension '711' priority 1 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '711' priority 2 to default [Feb 22 17:22:19] -- Added extension '711' priority 2 to default [Feb 22 17:22:19] -- Including context 'smvoice-intercom' in context 'default' [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_XXX' priority 1 to default [Feb 22 17:22:19] -- Added extension '_XXX' priority 1 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_XXX' priority 2 to default [Feb 22 17:22:19] -- Added extension '_XXX' priority 2 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_XXX' priority 3 to default [Feb 22 17:22:19] -- Added extension '_XXX' priority 3 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_XXX' priority 4 to default [Feb 22 17:22:19] -- Added extension '_XXX' priority 4 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_*XXX' priority 1 to default [Feb 22 17:22:19] -- Added extension '_*XXX' priority 1 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_*XXX' priority 2 to default [Feb 22 17:22:19] -- Added extension '_*XXX' priority 2 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_*XXX' priority 3 to default [Feb 22 17:22:19] -- Added extension '_*XXX' priority 3 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '7000' priority 1 to default [Feb 22 17:22:19] -- Added extension '7000' priority 1 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '7001' priority 1 to default [Feb 22 17:22:19] -- Added extension '7001' priority 1 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '7002' priority 1 to default [Feb 22 17:22:19] -- Added extension '7002' priority 1 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_XXXXX' priority 1 to default [Feb 22 17:22:19] -- Added extension '_XXXXX' priority 1 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_XXXXX' priority 2 to default [Feb 22 17:22:19] -- Added extension '_XXXXX' priority 2 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'i' priority 1 to default [Feb 22 17:22:19] -- Added extension 'i' priority 1 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'i' priority 2 to default [Feb 22 17:22:19] -- Added extension 'i' priority 2 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'i' priority 3 to default [Feb 22 17:22:19] -- Added extension 'i' priority 3 to default [Feb 22 17:22:19] DEBUG[14736]: pbx.c:3824 __ast_context_create: Registered context 'smvoice-local' [Feb 22 17:22:19] -- Registered extension context 'smvoice-local' [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '297' priority 1 to smvoice-local [Feb 22 17:22:19] -- Added extension '297' priority 1 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '298' priority 1 to smvoice-local [Feb 22 17:22:19] -- Added extension '298' priority 1 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '299' priority 1 to smvoice-local [Feb 22 17:22:19] -- Added extension '299' priority 1 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_1XX' priority 1 to smvoice-local [Feb 22 17:22:19] -- Added extension '_1XX' priority 1 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_1XX' priority 2 to smvoice-local [Feb 22 17:22:19] -- Added extension '_1XX' priority 2 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_1XX' priority 3 to smvoice-local [Feb 22 17:22:19] -- Added extension '_1XX' priority 3 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_1XX' priority 4 to smvoice-local [Feb 22 17:22:19] -- Added extension '_1XX' priority 4 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_1XX' priority 5 to smvoice-local [Feb 22 17:22:19] -- Added extension '_1XX' priority 5 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_1XX' priority 6 to smvoice-local [Feb 22 17:22:19] -- Added extension '_1XX' priority 6 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_1XX' priority 7 to smvoice-local [Feb 22 17:22:19] -- Added extension '_1XX' priority 7 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_1XX-CHANUNAVAIL' priority 1 to smvoice-local [Feb 22 17:22:19] -- Added extension '_1XX-CHANUNAVAIL' priority 1 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_1XX-CONGESTION' priority 1 to smvoice-local [Feb 22 17:22:19] -- Added extension '_1XX-CONGESTION' priority 1 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_1XX-NOANSWER' priority 1 to smvoice-local [Feb 22 17:22:19] -- Added extension '_1XX-NOANSWER' priority 1 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_1XX-BUSY' priority 1 to smvoice-local [Feb 22 17:22:19] -- Added extension '_1XX-BUSY' priority 1 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_206' priority 1 to smvoice-local [Feb 22 17:22:19] -- Added extension '_206' priority 1 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_2XX' priority 1 to smvoice-local [Feb 22 17:22:19] -- Added extension '_2XX' priority 1 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_2XX' priority 2 to smvoice-local [Feb 22 17:22:19] -- Added extension '_2XX' priority 2 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_2XX' priority 3 to smvoice-local [Feb 22 17:22:19] -- Added extension '_2XX' priority 3 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_2XX' priority 4 to smvoice-local [Feb 22 17:22:19] -- Added extension '_2XX' priority 4 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_2XX' priority 5 to smvoice-local [Feb 22 17:22:19] -- Added extension '_2XX' priority 5 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_2XX' priority 6 to smvoice-local [Feb 22 17:22:19] -- Added extension '_2XX' priority 6 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_2XX' priority 7 to smvoice-local [Feb 22 17:22:19] -- Added extension '_2XX' priority 7 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_2XX-CHANUNAVAIL' priority 1 to smvoice-local [Feb 22 17:22:19] -- Added extension '_2XX-CHANUNAVAIL' priority 1 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_2XX-CONGESTION' priority 1 to smvoice-local [Feb 22 17:22:19] -- Added extension '_2XX-CONGESTION' priority 1 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_2XX-NOANSWER' priority 1 to smvoice-local [Feb 22 17:22:19] -- Added extension '_2XX-NOANSWER' priority 1 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_2XX-BUSY' priority 1 to smvoice-local [Feb 22 17:22:19] -- Added extension '_2XX-BUSY' priority 1 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_4XX' priority 1 to smvoice-local [Feb 22 17:22:19] -- Added extension '_4XX' priority 1 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_4XX' priority 2 to smvoice-local [Feb 22 17:22:19] -- Added extension '_4XX' priority 2 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_4XX' priority 3 to smvoice-local [Feb 22 17:22:19] -- Added extension '_4XX' priority 3 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_4XX' priority 4 to smvoice-local [Feb 22 17:22:19] -- Added extension '_4XX' priority 4 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_4XX-CHANUNAVAIL' priority 1 to smvoice-local [Feb 22 17:22:19] -- Added extension '_4XX-CHANUNAVAIL' priority 1 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_4XX-CONGESTION' priority 1 to smvoice-local [Feb 22 17:22:19] -- Added extension '_4XX-CONGESTION' priority 1 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_4XX-NOANSWER' priority 1 to smvoice-local [Feb 22 17:22:19] -- Added extension '_4XX-NOANSWER' priority 1 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_4XX-BUSY' priority 1 to smvoice-local [Feb 22 17:22:19] -- Added extension '_4XX-BUSY' priority 1 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_5XX' priority 1 to smvoice-local [Feb 22 17:22:19] -- Added extension '_5XX' priority 1 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_5XX' priority 2 to smvoice-local [Feb 22 17:22:19] -- Added extension '_5XX' priority 2 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_5XX' priority 3 to smvoice-local [Feb 22 17:22:19] -- Added extension '_5XX' priority 3 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_5XX' priority 4 to smvoice-local [Feb 22 17:22:19] -- Added extension '_5XX' priority 4 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_5XX' priority 5 to smvoice-local [Feb 22 17:22:19] -- Added extension '_5XX' priority 5 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_5XX' priority 6 to smvoice-local [Feb 22 17:22:19] -- Added extension '_5XX' priority 6 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_5XX' priority 7 to smvoice-local [Feb 22 17:22:19] -- Added extension '_5XX' priority 7 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_5XX-CHANUNAVAIL' priority 1 to smvoice-local [Feb 22 17:22:19] -- Added extension '_5XX-CHANUNAVAIL' priority 1 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_5XX-CONGESTION' priority 1 to smvoice-local [Feb 22 17:22:19] -- Added extension '_5XX-CONGESTION' priority 1 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_5XX-NOANSWER' priority 1 to smvoice-local [Feb 22 17:22:19] -- Added extension '_5XX-NOANSWER' priority 1 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_5XX-BUSY' priority 1 to smvoice-local [Feb 22 17:22:19] -- Added extension '_5XX-BUSY' priority 1 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_6XX' priority 1 to smvoice-local [Feb 22 17:22:19] -- Added extension '_6XX' priority 1 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_6XX' priority 2 to smvoice-local [Feb 22 17:22:19] -- Added extension '_6XX' priority 2 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_6XX' priority 3 to smvoice-local [Feb 22 17:22:19] -- Added extension '_6XX' priority 3 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_6XX' priority 4 to smvoice-local [Feb 22 17:22:19] -- Added extension '_6XX' priority 4 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_6XX-CHANUNAVAIL' priority 1 to smvoice-local [Feb 22 17:22:19] -- Added extension '_6XX-CHANUNAVAIL' priority 1 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_6XX-CONGESTION' priority 1 to smvoice-local [Feb 22 17:22:19] -- Added extension '_6XX-CONGESTION' priority 1 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_6XX-NOANSWER' priority 1 to smvoice-local [Feb 22 17:22:19] -- Added extension '_6XX-NOANSWER' priority 1 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_6XX-BUSY' priority 1 to smvoice-local [Feb 22 17:22:19] -- Added extension '_6XX-BUSY' priority 1 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_8XX' priority 1 to smvoice-local [Feb 22 17:22:19] -- Added extension '_8XX' priority 1 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_8XX' priority 2 to smvoice-local [Feb 22 17:22:19] -- Added extension '_8XX' priority 2 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_8XX' priority 3 to smvoice-local [Feb 22 17:22:19] -- Added extension '_8XX' priority 3 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_8XX' priority 4 to smvoice-local [Feb 22 17:22:19] -- Added extension '_8XX' priority 4 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_8XX-CHANUNAVAIL' priority 1 to smvoice-local [Feb 22 17:22:19] -- Added extension '_8XX-CHANUNAVAIL' priority 1 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_8XX-CONGESTION' priority 1 to smvoice-local [Feb 22 17:22:19] -- Added extension '_8XX-CONGESTION' priority 1 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_8XX-NOANSWER' priority 1 to smvoice-local [Feb 22 17:22:19] -- Added extension '_8XX-NOANSWER' priority 1 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_8XX-BUSY' priority 1 to smvoice-local [Feb 22 17:22:19] -- Added extension '_8XX-BUSY' priority 1 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_*XXX' priority 1 to smvoice-local [Feb 22 17:22:19] -- Added extension '_*XXX' priority 1 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_*XXX' priority 2 to smvoice-local [Feb 22 17:22:19] -- Added extension '_*XXX' priority 2 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_*XXX' priority 3 to smvoice-local [Feb 22 17:22:19] -- Added extension '_*XXX' priority 3 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'INVALID' priority 1 to smvoice-local [Feb 22 17:22:19] -- Added extension 'INVALID' priority 1 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'INVALID' priority 2 to smvoice-local [Feb 22 17:22:19] -- Added extension 'INVALID' priority 2 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'INVALID' priority 3 to smvoice-local [Feb 22 17:22:19] -- Added extension 'INVALID' priority 3 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'i' priority 1 to smvoice-local [Feb 22 17:22:19] -- Added extension 'i' priority 1 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'i' priority 2 to smvoice-local [Feb 22 17:22:19] -- Added extension 'i' priority 2 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'i' priority 3 to smvoice-local [Feb 22 17:22:19] -- Added extension 'i' priority 3 to smvoice-local [Feb 22 17:22:19] DEBUG[14736]: pbx.c:3824 __ast_context_create: Registered context 'smvoice-sip' [Feb 22 17:22:19] -- Registered extension context 'smvoice-sip' [Feb 22 17:22:19] -- Including context 'smvoice-iaxy' in context 'smvoice-sip' [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '82473' priority 1 to smvoice-sip [Feb 22 17:22:19] -- Added extension '82473' priority 1 to smvoice-sip [Feb 22 17:22:19] DEBUG[14736]: pbx.c:3824 __ast_context_create: Registered context 'wellgate' [Feb 22 17:22:19] -- Registered extension context 'wellgate' [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 1 to wellgate [Feb 22 17:22:19] -- Added extension 's' priority 1 to wellgate [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '801' priority 1 to wellgate [Feb 22 17:22:19] -- Added extension '801' priority 1 to wellgate [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '801' priority 2 to wellgate [Feb 22 17:22:19] -- Added extension '801' priority 2 to wellgate [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '802' priority 1 to wellgate [Feb 22 17:22:19] -- Added extension '802' priority 1 to wellgate [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '802' priority 2 to wellgate [Feb 22 17:22:19] -- Added extension '802' priority 2 to wellgate [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '803' priority 1 to wellgate [Feb 22 17:22:19] -- Added extension '803' priority 1 to wellgate [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '803' priority 2 to wellgate [Feb 22 17:22:19] -- Added extension '803' priority 2 to wellgate [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '804' priority 1 to wellgate [Feb 22 17:22:19] -- Added extension '804' priority 1 to wellgate [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '804' priority 2 to wellgate [Feb 22 17:22:19] -- Added extension '804' priority 2 to wellgate [Feb 22 17:22:19] DEBUG[14736]: pbx.c:3824 __ast_context_create: Registered context 'smvoice-iaxy' [Feb 22 17:22:19] -- Registered extension context 'smvoice-iaxy' [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '50' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '50' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '50' priority 2 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '50' priority 2 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '55' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '55' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '56' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '56' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '57' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '57' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '57' priority 2 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '57' priority 2 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '57' priority 3 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '57' priority 3 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '57' priority 4 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '57' priority 4 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '58' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '58' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '58' priority 2 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '58' priority 2 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '58' priority 3 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '58' priority 3 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '58' priority 4 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '58' priority 4 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '59' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '59' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '5068012' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '5068012' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '199' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '199' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '1041' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '1041' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '1041' priority 2 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '1041' priority 2 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '1104' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '1104' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '1104' priority 2 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '1104' priority 2 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '10000' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '10000' priority 1 to smvoice-iaxy [Feb 22 17:22:19] WARNING[14736]: pbx.c:4618 add_pri: Unable to register extension '59', priority 1 in 'smvoice-iaxy', already in use [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '59' priority 2 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '59' priority 2 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '59' priority 3 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '59' priority 3 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '59' priority 4 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '59' priority 4 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '*70' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '*70' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '*71' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '*71' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '*72' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '*72' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '*73' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '*73' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '*74' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '*74' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '*75' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '*75' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '*76' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '*76' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '*77' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '*77' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '86' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '86' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '777' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '777' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '777' priority 2 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '777' priority 2 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '777' priority 3 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '777' priority 3 to smvoice-iaxy [Feb 22 17:22:19] -- Including context 'smvoice-intercom' in context 'smvoice-iaxy' [Feb 22 17:22:19] -- Including context 'smvoice-local' in context 'smvoice-iaxy' [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '7000' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '7000' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '7001' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '7001' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '7002' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '7002' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_9XXXXXXX' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_9XXXXXXX' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_9XXXXXXX' priority 2 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_9XXXXXXX' priority 2 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_9XXXXXXX-CHANUNAVAIL' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_9XXXXXXX-CHANUNAVAIL' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_9XXXXXXX-CHANUNAVAIL' priority 2 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_9XXXXXXX-CHANUNAVAIL' priority 2 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_9XXXXXXX-CONGESTION' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_9XXXXXXX-CONGESTION' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_9XXXXXXX-BUSY' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_9XXXXXXX-BUSY' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_9XXXXXXXXXX' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_9XXXXXXXXXX' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_9XXXXXXXXXX' priority 2 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_9XXXXXXXXXX' priority 2 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_9XXXXXXXXXX-CHANUNAVAIL' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_9XXXXXXXXXX-CHANUNAVAIL' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_9XXXXXXXXXX-CHANUNAVAIL' priority 2 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_9XXXXXXXXXX-CHANUNAVAIL' priority 2 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_9XXXXXXXXXX-CONGESTION' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_9XXXXXXXXXX-CONGESTION' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_9XXXXXXXXXX-BUSY' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_9XXXXXXXXXX-BUSY' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_91XXXXXXXXXX' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_91XXXXXXXXXX' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_91XXXXXXXXXX' priority 2 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_91XXXXXXXXXX' priority 2 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_91XXXXXXXXXX-CHANUNAVAIL' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_91XXXXXXXXXX-CHANUNAVAIL' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_91XXXXXXXXXX-CHANUNAVAIL' priority 2 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_91XXXXXXXXXX-CHANUNAVAIL' priority 2 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_91XXXXXXXXXX-CONGESTION' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_91XXXXXXXXXX-CONGESTION' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_91XXXXXXXXXX-BUSY' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_91XXXXXXXXXX-BUSY' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_8XXXXXXX' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_8XXXXXXX' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_8XXXXXXX' priority 2 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_8XXXXXXX' priority 2 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_8XXXXXXX' priority 3 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_8XXXXXXX' priority 3 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_8XXXXXXX-CHANUNAVAIL' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_8XXXXXXX-CHANUNAVAIL' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_8XXXXXXX-CHANUNAVAIL' priority 2 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_8XXXXXXX-CHANUNAVAIL' priority 2 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_8XXXXXXX-CONGESTION' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_8XXXXXXX-CONGESTION' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_8XXXXXXX-CONGESTION' priority 2 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_8XXXXXXX-CONGESTION' priority 2 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_8XXXXXXXXXX' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_8XXXXXXXXXX' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_8XXXXXXXXXX' priority 2 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_8XXXXXXXXXX' priority 2 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_8XXXXXXXXXX' priority 3 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_8XXXXXXXXXX' priority 3 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_8XXXXXXXXXX-CHANUNAVAIL' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_8XXXXXXXXXX-CHANUNAVAIL' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_8XXXXXXXXXX-CHANUNAVAIL' priority 2 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_8XXXXXXXXXX-CHANUNAVAIL' priority 2 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_8XXXXXXXXXX-CONGESTION' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_8XXXXXXXXXX-CONGESTION' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_8XXXXXXXXXX-CONGESTION' priority 2 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_8XXXXXXXXXX-CONGESTION' priority 2 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_81XXXXXXXXXX' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_81XXXXXXXXXX' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_81XXXXXXXXXX' priority 2 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_81XXXXXXXXXX' priority 2 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_81XXXXXXXXXX' priority 3 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_81XXXXXXXXXX' priority 3 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_81XXXXXXXXXX-CHANUNAVAIL' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_81XXXXXXXXXX-CHANUNAVAIL' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_81XXXXXXXXXX-CHANUNAVAIL' priority 2 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_81XXXXXXXXXX-CHANUNAVAIL' priority 2 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_81XXXXXXXXXX-CONGESTION' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_81XXXXXXXXXX-CONGESTION' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_81XXXXXXXXXX-CONGESTION' priority 2 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_81XXXXXXXXXX-CONGESTION' priority 2 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_7XXXXXXX' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_7XXXXXXX' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_7XXXXXXX' priority 2 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_7XXXXXXX' priority 2 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_7XXXXXXX' priority 3 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_7XXXXXXX' priority 3 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_71XXXXXXXXXX' priority 1 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_71XXXXXXXXXX' priority 1 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_7XXXXXXXXXXX' priority 2 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_7XXXXXXXXXXX' priority 2 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '_71XXXXXXXXXX' priority 3 to smvoice-iaxy [Feb 22 17:22:19] -- Added extension '_71XXXXXXXXXX' priority 3 to smvoice-iaxy [Feb 22 17:22:19] DEBUG[14736]: pbx.c:3824 __ast_context_create: Registered context 'smvoice-return-voicemail' [Feb 22 17:22:19] -- Registered extension context 'smvoice-return-voicemail' [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 1 to smvoice-return-voicemail [Feb 22 17:22:19] -- Added extension 's' priority 1 to smvoice-return-voicemail [Feb 22 17:22:19] DEBUG[14736]: pbx.c:3824 __ast_context_create: Registered context 'smvoice-voicemail' [Feb 22 17:22:19] -- Registered extension context 'smvoice-voicemail' [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '0' priority 1 to smvoice-voicemail [Feb 22 17:22:19] -- Added extension '0' priority 1 to smvoice-voicemail [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '1' priority 1 to smvoice-voicemail [Feb 22 17:22:19] -- Added extension '1' priority 1 to smvoice-voicemail [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '2' priority 1 to smvoice-voicemail [Feb 22 17:22:19] -- Added extension '2' priority 1 to smvoice-voicemail [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '3' priority 1 to smvoice-voicemail [Feb 22 17:22:19] -- Added extension '3' priority 1 to smvoice-voicemail [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '101' priority 1 to smvoice-voicemail [Feb 22 17:22:19] -- Added extension '101' priority 1 to smvoice-voicemail [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '510' priority 1 to smvoice-voicemail [Feb 22 17:22:19] -- Added extension '510' priority 1 to smvoice-voicemail [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '511' priority 1 to smvoice-voicemail [Feb 22 17:22:19] -- Added extension '511' priority 1 to smvoice-voicemail [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '512' priority 1 to smvoice-voicemail [Feb 22 17:22:19] -- Added extension '512' priority 1 to smvoice-voicemail [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '513' priority 1 to smvoice-voicemail [Feb 22 17:22:19] -- Added extension '513' priority 1 to smvoice-voicemail [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '514' priority 1 to smvoice-voicemail [Feb 22 17:22:19] -- Added extension '514' priority 1 to smvoice-voicemail [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '515' priority 1 to smvoice-voicemail [Feb 22 17:22:19] -- Added extension '515' priority 1 to smvoice-voicemail [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '806' priority 1 to smvoice-voicemail [Feb 22 17:22:19] -- Added extension '806' priority 1 to smvoice-voicemail [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '205' priority 1 to smvoice-voicemail [Feb 22 17:22:19] -- Added extension '205' priority 1 to smvoice-voicemail [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '522' priority 1 to smvoice-voicemail [Feb 22 17:22:19] -- Added extension '522' priority 1 to smvoice-voicemail [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '605' priority 1 to smvoice-voicemail [Feb 22 17:22:19] -- Added extension '605' priority 1 to smvoice-voicemail [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '801' priority 1 to smvoice-voicemail [Feb 22 17:22:19] -- Added extension '801' priority 1 to smvoice-voicemail [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '210' priority 1 to smvoice-voicemail [Feb 22 17:22:19] -- Added extension '210' priority 1 to smvoice-voicemail [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '216' priority 1 to smvoice-voicemail [Feb 22 17:22:19] -- Added extension '216' priority 1 to smvoice-voicemail [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '592' priority 1 to smvoice-voicemail [Feb 22 17:22:19] -- Added extension '592' priority 1 to smvoice-voicemail [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '2134' priority 1 to smvoice-voicemail [Feb 22 17:22:19] -- Added extension '2134' priority 1 to smvoice-voicemail [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '209' priority 1 to smvoice-voicemail [Feb 22 17:22:19] -- Added extension '209' priority 1 to smvoice-voicemail [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '204' priority 1 to smvoice-voicemail [Feb 22 17:22:19] -- Added extension '204' priority 1 to smvoice-voicemail [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '401' priority 1 to smvoice-voicemail [Feb 22 17:22:19] -- Added extension '401' priority 1 to smvoice-voicemail [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '402' priority 1 to smvoice-voicemail [Feb 22 17:22:19] -- Added extension '402' priority 1 to smvoice-voicemail [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '403' priority 1 to smvoice-voicemail [Feb 22 17:22:19] -- Added extension '403' priority 1 to smvoice-voicemail [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '404' priority 1 to smvoice-voicemail [Feb 22 17:22:19] -- Added extension '404' priority 1 to smvoice-voicemail [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '528' priority 1 to smvoice-voicemail [Feb 22 17:22:19] -- Added extension '528' priority 1 to smvoice-voicemail [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '530' priority 1 to smvoice-voicemail [Feb 22 17:22:19] -- Added extension '530' priority 1 to smvoice-voicemail [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '531' priority 1 to smvoice-voicemail [Feb 22 17:22:19] -- Added extension '531' priority 1 to smvoice-voicemail [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '606' priority 1 to smvoice-voicemail [Feb 22 17:22:19] -- Added extension '606' priority 1 to smvoice-voicemail [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '616' priority 1 to smvoice-voicemail [Feb 22 17:22:19] -- Added extension '616' priority 1 to smvoice-voicemail [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '800' priority 1 to smvoice-voicemail [Feb 22 17:22:19] -- Added extension '800' priority 1 to smvoice-voicemail [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'i' priority 1 to smvoice-voicemail [Feb 22 17:22:19] -- Added extension 'i' priority 1 to smvoice-voicemail [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'i' priority 2 to smvoice-voicemail [Feb 22 17:22:19] -- Added extension 'i' priority 2 to smvoice-voicemail [Feb 22 17:22:19] DEBUG[14736]: pbx.c:3824 __ast_context_create: Registered context 'smvoice-intercom' [Feb 22 17:22:19] -- Registered extension context 'smvoice-intercom' [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '87' priority 1 to smvoice-intercom [Feb 22 17:22:19] -- Added extension '87' priority 1 to smvoice-intercom [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension '87' priority 2 to smvoice-intercom [Feb 22 17:22:19] -- Added extension '87' priority 2 to smvoice-intercom [Feb 22 17:22:19] DEBUG[14736]: pbx.c:3824 __ast_context_create: Registered context 'smvoice-incoming' [Feb 22 17:22:19] -- Registered extension context 'smvoice-incoming' [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 1 to smvoice-incoming [Feb 22 17:22:19] -- Added extension 's' priority 1 to smvoice-incoming [Feb 22 17:22:19] -- Including context 'smvoice-intercom' in context 'smvoice-incoming' [Feb 22 17:22:19] -- Including context 'smvoice-transfers' in context 'smvoice-incoming' [Feb 22 17:22:19] DEBUG[14736]: pbx.c:3824 __ast_context_create: Registered context 'smvoice-incoming-kx1232' [Feb 22 17:22:19] -- Registered extension context 'smvoice-incoming-kx1232' [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 1 to smvoice-incoming-kx1232 [Feb 22 17:22:19] -- Added extension 's' priority 1 to smvoice-incoming-kx1232 [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 2 to smvoice-incoming-kx1232 [Feb 22 17:22:19] -- Added extension 's' priority 2 to smvoice-incoming-kx1232 [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 3 to smvoice-incoming-kx1232 [Feb 22 17:22:19] -- Added extension 's' priority 3 to smvoice-incoming-kx1232 [Feb 22 17:22:19] DEBUG[14736]: pbx.c:3824 __ast_context_create: Registered context 'smvoice-transfers' [Feb 22 17:22:19] -- Registered extension context 'smvoice-transfers' [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'smvoice_dial' priority 1 to smvoice-transfers [Feb 22 17:22:19] -- Added extension 'smvoice_dial' priority 1 to smvoice-transfers [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'smvoice_dial' priority 2 to smvoice-transfers [Feb 22 17:22:19] -- Added extension 'smvoice_dial' priority 2 to smvoice-transfers [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'smvoice_dial-CHANUNAVAIL' priority 1 to smvoice-transfers [Feb 22 17:22:19] -- Added extension 'smvoice_dial-CHANUNAVAIL' priority 1 to smvoice-transfers [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'smvoice_dial-CHANUNAVAIL' priority 2 to smvoice-transfers [Feb 22 17:22:19] -- Added extension 'smvoice_dial-CHANUNAVAIL' priority 2 to smvoice-transfers [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'smvoice_dial-CONGESTION' priority 1 to smvoice-transfers [Feb 22 17:22:19] -- Added extension 'smvoice_dial-CONGESTION' priority 1 to smvoice-transfers [Feb 22 17:22:19] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 42fcb9a2205a226e1657915a20f1c05a@24.123.23.170 - REGISTER (No RTP) [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'smvoice_dial-CONGESTION' priority 2 to smvoice-transfers [Feb 22 17:22:19] -- Added extension 'smvoice_dial-CONGESTION' priority 2 to smvoice-transfers [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'smvoice_dial-NOANSWER' priority 1 to smvoice-transfers [Feb 22 17:22:19] -- Added extension 'smvoice_dial-NOANSWER' priority 1 to smvoice-transfers [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'smvoice_dial-NOANSWER' priority 2 to smvoice-transfers [Feb 22 17:22:19] -- Added extension 'smvoice_dial-NOANSWER' priority 2 to smvoice-transfers [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'smvoice_dial-BUSY' priority 1 to smvoice-transfers [Feb 22 17:22:19] -- Added extension 'smvoice_dial-BUSY' priority 1 to smvoice-transfers [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'smvoice_dial-BUSY' priority 2 to smvoice-transfers [Feb 22 17:22:19] -- Added extension 'smvoice_dial-BUSY' priority 2 to smvoice-transfers [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'smvoice_dial_no_extension' priority 1 to smvoice-transfers [Feb 22 17:22:19] -- Added extension 'smvoice_dial_no_extension' priority 1 to smvoice-transfers [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'smvoice_dial_no_extension' priority 2 to smvoice-transfers [Feb 22 17:22:19] -- Added extension 'smvoice_dial_no_extension' priority 2 to smvoice-transfers [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'smvoice_dial_no_extension-CHANUNAVAIL' priority 1 to smvoice-transfers [Feb 22 17:22:19] -- Added extension 'smvoice_dial_no_extension-CHANUNAVAIL' priority 1 to smvoice-transfers [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'smvoice_dial_no_extension-CHANUNAVAIL' priority 2 to smvoice-transfers [Feb 22 17:22:19] -- Added extension 'smvoice_dial_no_extension-CHANUNAVAIL' priority 2 to smvoice-transfers [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'smvoice_dial_no_extension-CONGESTION' priority 1 to smvoice-transfers [Feb 22 17:22:19] -- Added extension 'smvoice_dial_no_extension-CONGESTION' priority 1 to smvoice-transfers [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'smvoice_dial_no_extension-BUSY' priority 1 to smvoice-transfers [Feb 22 17:22:19] -- Added extension 'smvoice_dial_no_extension-BUSY' priority 1 to smvoice-transfers [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'smvoice_dial_no_extension-NOANSWER' priority 1 to smvoice-transfers [Feb 22 17:22:19] -- Added extension 'smvoice_dial_no_extension-NOANSWER' priority 1 to smvoice-transfers [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'smvoice_dial_no_extension-NOANSWER' priority 2 to smvoice-transfers [Feb 22 17:22:19] -- Added extension 'smvoice_dial_no_extension-NOANSWER' priority 2 to smvoice-transfers [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'smvoice_dial_goto_voicemail' priority 1 to smvoice-transfers [Feb 22 17:22:19] -- Added extension 'smvoice_dial_goto_voicemail' priority 1 to smvoice-transfers [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'smvoice_dial_goto_voicemail' priority 2 to smvoice-transfers [Feb 22 17:22:19] -- Added extension 'smvoice_dial_goto_voicemail' priority 2 to smvoice-transfers [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'smvoice_dial_goto_voicemail-CHANUNAVAIL' priority 1 to smvoice-transfers [Feb 22 17:22:19] -- Added extension 'smvoice_dial_goto_voicemail-CHANUNAVAIL' priority 1 to smvoice-transfers [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'smvoice_dial_goto_voicemail-CONGESTION' priority 1 to smvoice-transfers [Feb 22 17:22:19] -- Added extension 'smvoice_dial_goto_voicemail-CONGESTION' priority 1 to smvoice-transfers [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'smvoice_dial_goto_voicemail-NOANSWER' priority 1 to smvoice-transfers [Feb 22 17:22:19] -- Added extension 'smvoice_dial_goto_voicemail-NOANSWER' priority 1 to smvoice-transfers [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'smvoice_dial_goto_voicemail-BUSY' priority 1 to smvoice-transfers [Feb 22 17:22:19] -- Added extension 'smvoice_dial_goto_voicemail-BUSY' priority 1 to smvoice-transfers [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'smvoice_conference' priority 1 to smvoice-transfers [Feb 22 17:22:19] -- Added extension 'smvoice_conference' priority 1 to smvoice-transfers [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'smvoice_sendtext' priority 1 to smvoice-transfers [Feb 22 17:22:19] -- Added extension 'smvoice_sendtext' priority 1 to smvoice-transfers [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'smvoice_sendtext' priority 2 to smvoice-transfers [Feb 22 17:22:19] -- Added extension 'smvoice_sendtext' priority 2 to smvoice-transfers [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'smvoice_sendtext' priority 3 to smvoice-transfers [Feb 22 17:22:19] -- Added extension 'smvoice_sendtext' priority 3 to smvoice-transfers [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'smvoice_no_callprogress' priority 1 to smvoice-transfers [Feb 22 17:22:19] -- Added extension 'smvoice_no_callprogress' priority 1 to smvoice-transfers [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'smvoice_callprogress' priority 1 to smvoice-transfers [Feb 22 17:22:19] -- Added extension 'smvoice_callprogress' priority 1 to smvoice-transfers [Feb 22 17:22:19] DEBUG[14736]: pbx.c:3824 __ast_context_create: Registered context 'smvoice-homework-hotline' [Feb 22 17:22:19] -- Registered extension context 'smvoice-homework-hotline' [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 1 to smvoice-homework-hotline [Feb 22 17:22:19] -- Added extension 's' priority 1 to smvoice-homework-hotline [Feb 22 17:22:19] DEBUG[14736]: pbx.c:3824 __ast_context_create: Registered context 'smvoice-faxout' [Feb 22 17:22:19] -- Registered extension context 'smvoice-faxout' [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'smvoice_faxout' priority 1 to smvoice-faxout [Feb 22 17:22:19] -- Added extension 'smvoice_faxout' priority 1 to smvoice-faxout [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'smvoice_faxout' priority 2 to smvoice-faxout [Feb 22 17:22:19] -- Added extension 'smvoice_faxout' priority 2 to smvoice-faxout [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'smvoice_faxout' priority 3 to smvoice-faxout [Feb 22 17:22:19] -- Added extension 'smvoice_faxout' priority 3 to smvoice-faxout [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'smvoice_faxout' priority 4 to smvoice-faxout [Feb 22 17:22:19] -- Added extension 'smvoice_faxout' priority 4 to smvoice-faxout [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'smvoice_faxout' priority 104 to smvoice-faxout [Feb 22 17:22:19] -- Added extension 'smvoice_faxout' priority 104 to smvoice-faxout [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'smvoice_faxout' priority 105 to smvoice-faxout [Feb 22 17:22:19] -- Added extension 'smvoice_faxout' priority 105 to smvoice-faxout [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'i' priority 1 to smvoice-faxout [Feb 22 17:22:19] -- Added extension 'i' priority 1 to smvoice-faxout [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'i' priority 2 to smvoice-faxout [Feb 22 17:22:19] -- Added extension 'i' priority 2 to smvoice-faxout [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 't' priority 1 to smvoice-faxout [Feb 22 17:22:19] -- Added extension 't' priority 1 to smvoice-faxout [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 't' priority 2 to smvoice-faxout [Feb 22 17:22:19] -- Added extension 't' priority 2 to smvoice-faxout [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'T' priority 1 to smvoice-faxout [Feb 22 17:22:19] -- Added extension 'T' priority 1 to smvoice-faxout [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'T' priority 2 to smvoice-faxout [Feb 22 17:22:19] -- Added extension 'T' priority 2 to smvoice-faxout [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'failed' priority 1 to smvoice-faxout [Feb 22 17:22:19] -- Added extension 'failed' priority 1 to smvoice-faxout [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'failed' priority 2 to smvoice-faxout [Feb 22 17:22:19] -- Added extension 'failed' priority 2 to smvoice-faxout [Feb 22 17:22:19] DEBUG[14736]: pbx.c:3824 __ast_context_create: Registered context 'smvoice-dialout' [Feb 22 17:22:19] -- Registered extension context 'smvoice-dialout' [Feb 22 17:22:19] -- Including context 'smvoice-intercom' in context 'smvoice-dialout' [Feb 22 17:22:19] -- Including context 'smvoice-transfers' in context 'smvoice-dialout' [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'i' priority 1 to smvoice-dialout [Feb 22 17:22:19] -- Added extension 'i' priority 1 to smvoice-dialout [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 't' priority 1 to smvoice-dialout [Feb 22 17:22:19] -- Added extension 't' priority 1 to smvoice-dialout [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'T' priority 1 to smvoice-dialout [Feb 22 17:22:19] -- Added extension 'T' priority 1 to smvoice-dialout [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'failed' priority 1 to smvoice-dialout [Feb 22 17:22:19] -- Added extension 'failed' priority 1 to smvoice-dialout [Feb 22 17:22:19] DEBUG[14736]: pbx.c:3824 __ast_context_create: Registered context 'iax_devcentos64_to_unifiedpaging' [Feb 22 17:22:19] -- Registered extension context 'iax_devcentos64_to_unifiedpaging' [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 's' priority 1 to iax_devcentos64_to_unifiedpaging [Feb 22 17:22:19] -- Added extension 's' priority 1 to iax_devcentos64_to_unifiedpaging [Feb 22 17:22:19] DEBUG[14736]: pbx.c:3824 __ast_context_create: Registered context 'smvoice-testing' [Feb 22 17:22:19] -- Registered extension context 'smvoice-testing' [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'call_cell' priority 1 to smvoice-testing [Feb 22 17:22:19] -- Added extension 'call_cell' priority 1 to smvoice-testing [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'call_cell' priority 2 to smvoice-testing [Feb 22 17:22:19] -- Added extension 'call_cell' priority 2 to smvoice-testing [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'call_cell' priority 3 to smvoice-testing [Feb 22 17:22:19] -- Added extension 'call_cell' priority 3 to smvoice-testing [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'call_cell' priority 4 to smvoice-testing [Feb 22 17:22:19] -- Added extension 'call_cell' priority 4 to smvoice-testing [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'failed' priority 1 to smvoice-testing [Feb 22 17:22:19] -- Added extension 'failed' priority 1 to smvoice-testing [Feb 22 17:22:19] DEBUG[14736]: pbx.c:4794 ast_add_extension2: Added extension 'failed' priority 2 to smvoice-testing [Feb 22 17:22:19] -- Added extension 'failed' priority 2 to smvoice-testing [Feb 22 17:22:19] DEBUG[14736]: pbx.c:3898 ast_merge_contexts_and_delete: must remove any reg pbx_config [Feb 22 17:22:19] DEBUG[14736]: pbx.c:5231 __ast_context_destroy: check ctx ael-demo pbx_ael [Feb 22 17:22:19] DEBUG[14736]: pbx.c:5231 __ast_context_destroy: check ctx macro-std-exten-ael pbx_ael [Feb 22 17:22:19] DEBUG[14736]: pbx.c:5231 __ast_context_destroy: check ctx parkedcalls res_features [Feb 22 17:22:19] pbx_config.so => (Text Extension Configuration) [Feb 22 17:22:19] == Registered file format wav49, extension(s) WAV|wav49 [Feb 22 17:22:19] format_wav_gsm.so => (Microsoft WAV format (Proprietary GSM)) [Feb 22 17:22:19] == Registered application 'LookupCIDName' [Feb 22 17:22:19] app_lookupcidname.so => (Look up CallerID Name from local database) [Feb 22 17:22:19] == Registered application 'Directory' [Feb 22 17:22:19] app_directory.so => (Extension Directory) [Feb 22 17:22:19] == Registered application 'Flash' [Feb 22 17:22:19] app_flash.so => (Flash channel application) [Feb 22 17:22:19] == Registered application 'ZapScan' [Feb 22 17:22:19] app_zapscan.so => (Scan Zap channels application) [Feb 22 17:22:19] == Registered file format iLBC, extension(s) ilbc [Feb 22 17:22:19] format_ilbc.so => (Raw iLBC data) [Feb 22 17:22:19] DEBUG[14736]: channel.c:449 ast_channel_register: Registered handler for 'Agent' (Call Agent Proxy Channel) [Feb 22 17:22:19] == Registered channel type 'Agent' (Call Agent Proxy Channel) [Feb 22 17:22:19] == Parsing '/etc/asterisk/agents.conf': [Feb 22 17:22:19] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/agents.conf [Feb 22 17:22:19] Found [Feb 22 17:22:19] == Registered application 'AgentLogin' [Feb 22 17:22:19] == Registered application 'AgentCallbackLogin' [Feb 22 17:22:19] == Registered application 'AgentMonitorOutgoing' [Feb 22 17:22:19] == Manager registered action Agents [Feb 22 17:22:19] == Manager registered action AgentLogoff [Feb 22 17:22:19] == Manager registered action AgentCallbackLogin [Feb 22 17:22:19] == Registered custom function AGENT [Feb 22 17:22:19] chan_agent.so => (Agent Proxy Channel) [Feb 22 17:22:19] == Registered application 'SetCallerPres' [Feb 22 17:22:19] == Registered application 'SetCallerID' [Feb 22 17:22:19] app_setcallerid.so => (Set CallerID Application) [Feb 22 17:22:19] == Registered application 'Transfer' [Feb 22 17:22:19] app_transfer.so => (Transfer) [Feb 22 17:22:19] DEBUG[14736]: channel.c:449 ast_channel_register: Registered handler for 'Local' (Local Proxy Channel Driver) [Feb 22 17:22:19] == Registered channel type 'Local' (Local Proxy Channel Driver) [Feb 22 17:22:19] chan_local.so => (Local Proxy Channel) [Feb 22 17:22:19] == Parsing '/etc/asterisk/codecs.conf': [Feb 22 17:22:19] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/codecs.conf [Feb 22 17:22:19] Found [Feb 22 17:22:19] -- codec_g726: using generic PLC [Feb 22 17:22:19] WARNING[14736]: translate.c:675 __ast_register_translator: plc_samples 160 format 6 [Feb 22 17:22:19] == Registered translator 'g726tolin' from format g726 to slin, cost 1 [Feb 22 17:22:19] DEBUG[14736]: translate.c:425 rebuild_matrix: Resetting translation matrix [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] == Registered translator 'lintog726' from format slin to g726, cost 1 [Feb 22 17:22:19] DEBUG[14736]: translate.c:425 rebuild_matrix: Resetting translation matrix [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] WARNING[14736]: translate.c:675 __ast_register_translator: plc_samples 160 format 6 [Feb 22 17:22:19] == Registered translator 'g726aal2tolin' from format g726aal2 to slin, cost 1 [Feb 22 17:22:19] DEBUG[14736]: translate.c:425 rebuild_matrix: Resetting translation matrix [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from ulaw to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from ulaw to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from ulaw to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from ulaw to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from ulaw to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14767]: chan_sip.c:7214 transmit_register: Scheduled a registration timeout for sip.broadvoice.com id #80 [Feb 22 17:22:19] REGISTER attempt 1 to 3173241052@sip.broadvoice.com [Feb 22 17:22:19] == Registered translator 'lintog726aal2' from format slin to g726aal2, cost 2 [Feb 22 17:22:19] DEBUG[14736]: translate.c:425 rebuild_matrix: Resetting translation matrix [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from g723 to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from ulaw to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from ulaw to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from ulaw to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from ulaw to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from ulaw to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] == Registered translator 'g726aal2tog726' from format g726aal2 to g726, cost 1 [Feb 22 17:22:19] DEBUG[14736]: translate.c:425 rebuild_matrix: Resetting translation matrix [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from g723 to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from ulaw to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from ulaw to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from ulaw to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from ulaw to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] == Registered translator 'g726tog726aal2' from format g726 to g726aal2, cost 1 [Feb 22 17:22:19] DEBUG[14736]: translate.c:425 rebuild_matrix: Resetting translation matrix [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from g723 to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from ulaw to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from ulaw to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from ulaw to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from ulaw to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 4 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 4 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 4 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 4 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] codec_g726.so => (ITU G.726-32kbps G726 Transcoder) [Feb 22 17:22:19] == Registered application 'ZapSendKeypadFacility' [Feb 22 17:22:19] == Parsing '/etc/asterisk/zapata.conf': [Feb 22 17:22:19] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/zapata.conf [Feb 22 17:22:19] Found [Feb 22 17:22:19] DEBUG[14736]: chan_zap.c:1430 update_conf: Updated conferencing on 1, with 0 conference users [Feb 22 17:22:19] -- Registered channel 1, FXS Kewlstart signalling [Feb 22 17:22:19] DEBUG[14736]: chan_zap.c:1430 update_conf: Updated conferencing on 2, with 0 conference users [Feb 22 17:22:19] -- Registered channel 2, FXS Kewlstart signalling [Feb 22 17:22:19] DEBUG[14736]: chan_zap.c:1430 update_conf: Updated conferencing on 3, with 0 conference users [Feb 22 17:22:19] -- Registered channel 3, FXS Kewlstart signalling [Feb 22 17:22:19] DEBUG[14736]: chan_zap.c:1430 update_conf: Updated conferencing on 4, with 0 conference users [Feb 22 17:22:19] -- Registered channel 4, FXS Kewlstart signalling [Feb 22 17:22:19] DEBUG[14736]: chan_zap.c:1430 update_conf: Updated conferencing on 5, with 0 conference users [Feb 22 17:22:19] -- Registered channel 5, FXS Kewlstart signalling [Feb 22 17:22:19] DEBUG[14736]: chan_zap.c:1430 update_conf: Updated conferencing on 6, with 0 conference users [Feb 22 17:22:19] -- Registered channel 6, FXS Kewlstart signalling [Feb 22 17:22:19] DEBUG[14736]: chan_zap.c:1430 update_conf: Updated conferencing on 7, with 0 conference users [Feb 22 17:22:19] -- Registered channel 7, FXS Kewlstart signalling [Feb 22 17:22:19] DEBUG[14736]: chan_zap.c:1430 update_conf: Updated conferencing on 8, with 0 conference users [Feb 22 17:22:19] -- Registered channel 8, FXS Kewlstart signalling [Feb 22 17:22:19] -- Automatically generated pseudo channel [Feb 22 17:22:19] DEBUG[14736]: channel.c:449 ast_channel_register: Registered handler for 'Zap' (Zapata Telephony Driver w/PRI) [Feb 22 17:22:19] == Registered channel type 'Zap' (Zapata Telephony Driver w/PRI) [Feb 22 17:22:19] == Manager registered action ZapTransfer [Feb 22 17:22:19] == Manager registered action ZapHangup [Feb 22 17:22:19] == Manager registered action ZapDialOffhook [Feb 22 17:22:19] == Manager registered action ZapDNDon [Feb 22 17:22:19] == Manager registered action ZapDNDoff [Feb 22 17:22:19] == Manager registered action ZapShowChannels [Feb 22 17:22:19] == Manager registered action ZapRestart [Feb 22 17:22:19] chan_zap.so => (Zapata Telephony) [Feb 22 17:22:19] == Registered application 'ControlPlayback' [Feb 22 17:22:19] app_controlplayback.so => (Control Playback Application) [Feb 22 17:22:19] == Registered custom function URIDECODE [Feb 22 17:22:19] == Registered custom function URIENCODE [Feb 22 17:22:19] func_uri.so => (URI encode/decode dialplan functions) [Feb 22 17:22:19] == Registered custom function FIELDQTY [Feb 22 17:22:19] == Registered custom function FILTER [Feb 22 17:22:19] == Registered custom function REGEX [Feb 22 17:22:19] == Registered custom function ARRAY [Feb 22 17:22:19] == Registered custom function QUOTE [Feb 22 17:22:19] == Registered custom function LEN [Feb 22 17:22:19] == Registered custom function STRFTIME [Feb 22 17:22:19] == Registered custom function STRPTIME [Feb 22 17:22:19] == Registered custom function EVAL [Feb 22 17:22:19] == Registered custom function KEYPADHASH [Feb 22 17:22:19] == Registered custom function SPRINTF [Feb 22 17:22:19] func_strings.so => (String handling dialplan functions) [Feb 22 17:22:19] == Registered application 'ChanSpy' [Feb 22 17:22:19] == Registered application 'ExtenSpy' [Feb 22 17:22:19] app_chanspy.so => (Listen to the audio of an active channel) [Feb 22 17:22:19] == Registered application 'SetTransferCapability' [Feb 22 17:22:19] app_settransfercapability.so => (Set ISDN Transfer Capability) [Feb 22 17:22:19] == Registered application 'Pickup' [Feb 22 17:22:19] app_directed_pickup.so => (Directed Call Pickup Application) [Feb 22 17:22:19] == Registered application 'NBScat' [Feb 22 17:22:19] app_nbscat.so => (Silly NBS Stream Application) [Feb 22 17:22:19] == Registered custom function TIMEOUT [Feb 22 17:22:19] func_timeout.so => (Channel timeout dialplan functions) [Feb 22 17:22:19] == Registered custom function CALLERID [Feb 22 17:22:19] func_callerid.so => (Caller ID related dialplan function) [Feb 22 17:22:19] == Registered application 'VoiceMail' [Feb 22 17:22:19] == Registered application 'VoiceMailMain' [Feb 22 17:22:19] == Registered application 'MailboxExists' [Feb 22 17:22:19] == Registered application 'VMAuthenticate' [Feb 22 17:22:19] == Parsing '/etc/asterisk/voicemail.conf': [Feb 22 17:22:19] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/voicemail.conf [Feb 22 17:22:19] Found [Feb 22 17:22:19] DEBUG[14736]: app_voicemail.c:7250 load_config: VM Review Option disabled globally [Feb 22 17:22:19] DEBUG[14736]: app_voicemail.c:7257 load_config: VM Temperary Greeting Reminder Option disabled globally [Feb 22 17:22:19] DEBUG[14736]: app_voicemail.c:7265 load_config: VM Operator break disabled globally [Feb 22 17:22:19] DEBUG[14736]: app_voicemail.c:7271 load_config: VM CID Info before msg disabled globally [Feb 22 17:22:19] DEBUG[14736]: app_voicemail.c:7283 load_config: ENVELOPE before msg enabled globally [Feb 22 17:22:19] DEBUG[14736]: app_voicemail.c:7289 load_config: Duration info before msg enabled globally [Feb 22 17:22:19] DEBUG[14736]: app_voicemail.c:7304 load_config: We are not going to skip to the next msg after save/delete [Feb 22 17:22:19] app_voicemail.so => (Comedian Mail (Voicemail System)) [Feb 22 17:22:19] == Registered application 'ExternalIVR' [Feb 22 17:22:19] app_externalivr.so => (External IVR Interface Application) [Feb 22 17:22:19] == Registered custom function CHANNEL [Feb 22 17:22:19] func_channel.so => (Channel information dialplan function) [Feb 22 17:22:19] == Registered application 'MP3Player' [Feb 22 17:22:19] app_mp3.so => (Silly MP3 Application) [Feb 22 17:22:19] == Registered application 'ReadFile' [Feb 22 17:22:19] app_readfile.so => (Stores output of file into a variable) [Feb 22 17:22:19] == Registered application 'SendImage' [Feb 22 17:22:19] app_image.so => (Image Transmission Application) [Feb 22 17:22:19] == Registered application 'While' [Feb 22 17:22:19] == Registered application 'EndWhile' [Feb 22 17:22:19] == Registered application 'ExitWhile' [Feb 22 17:22:19] == Registered application 'ContinueWhile' [Feb 22 17:22:19] app_while.so => (While Loops and Conditional Execution) [Feb 22 17:22:19] == Registered application 'SayUnixTime' [Feb 22 17:22:19] == Registered application 'DateTime' [Feb 22 17:22:19] app_sayunixtime.so => (Say time) [Feb 22 17:22:19] == Registered custom function BASE64_ENCODE [Feb 22 17:22:19] == Registered custom function BASE64_DECODE [Feb 22 17:22:19] func_base64.so => (base64 encode/decode dialplan functions) [Feb 22 17:22:19] ERROR[14736]: app_amd.c:329 load_config: Configuration file amd.conf missing. [Feb 22 17:22:19] == Registered application 'AMD' [Feb 22 17:22:19] app_amd.so => (Answering Machine Detection Application) [Feb 22 17:22:19] == Registered application 'WaitForRing' [Feb 22 17:22:19] app_waitforring.so => (Waits until first ring after time) [Feb 22 17:22:19] == Registered custom function LANGUAGE [Feb 22 17:22:19] func_language.so => (Channel language dialplan function) [Feb 22 17:22:19] == Registered application 'Zapateller' [Feb 22 17:22:19] app_zapateller.so => (Block Telemarketers with Special Information Tone) [Feb 22 17:22:19] == Registered application 'NoCDR' [Feb 22 17:22:19] app_cdr.so => (Tell Asterisk to not maintain a CDR for the current call) [Feb 22 17:22:19] == Registered application 'Playback' [Feb 22 17:22:19] app_playback.so => (Sound File Playback Application) [Feb 22 17:22:19] == Manager registered action PlayDTMF [Feb 22 17:22:19] == Registered application 'SendDTMF' [Feb 22 17:22:19] app_senddtmf.so => (Send DTMF digits Application) [Feb 22 17:22:19] == Registered application 'ForkCDR' [Feb 22 17:22:19] app_forkcdr.so => (Fork The CDR into 2 separate entities) [Feb 22 17:22:19] == Registered application 'UserEvent' [Feb 22 17:22:19] app_userevent.so => (Custom User Event Application) [Feb 22 17:22:19] == Registered application 'TrySystem' [Feb 22 17:22:19] == Registered application 'System' [Feb 22 17:22:19] app_system.so => (Generic System() application) [Feb 22 17:22:19] == Registered translator 'alawtoulaw' from format alaw to ulaw, cost 1 [Feb 22 17:22:19] DEBUG[14736]: translate.c:425 rebuild_matrix: Resetting translation matrix [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from g723 to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from ulaw to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from ulaw to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from ulaw to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from ulaw to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to gsm, via 3 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to gsm, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to gsm, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 4 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 4 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 4 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 4 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to gsm, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from g723 to gsm, via 3 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 7 cost path from ulaw to gsm, via 1 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from ulaw to gsm, via 3 [Feb 22 17:22:19] == Registered translator 'ulawtoalaw' from format ulaw to alaw, cost 1 [Feb 22 17:22:19] DEBUG[14736]: translate.c:425 rebuild_matrix: Resetting translation matrix [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from g723 to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from gsm to unknown, via 3 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from gsm to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from gsm to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from gsm to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 16 cost path from gsm to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from gsm to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 6 cost path from unknown to g723, via 2 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to ulaw, via 2 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 6 cost path from unknown to unknown, via 2 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 17 cost path from unknown to unknown, via 2 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 2 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from ulaw to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from ulaw to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from ulaw to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from ulaw to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to gsm, via 3 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to gsm, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to gsm, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14767]: chan_sip.c:4315 find_call: = Found Their Call ID: 42fcb9a2205a226e1657915a20f1c05a@24.123.23.170 Their Tag Our tag: as5416a864 [Feb 22 17:22:19] DEBUG[14767]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '42fcb9a2205a226e1657915a20f1c05a@24.123.23.170' of Request 102: Match Not Found [Feb 22 17:22:19] DEBUG[14767]: chan_sip.c:11906 handle_response_register: Registration successful [Feb 22 17:22:19] DEBUG[14767]: chan_sip.c:11909 handle_response_register: Cancelling timeout 80 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 4 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 4 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 4 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 4 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to gsm, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from g723 to gsm, via 3 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 7 cost path from ulaw to gsm, via 1 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from ulaw to gsm, via 3 [Feb 22 17:22:19] codec_a_mu.so => (A-law and Mulaw direct Coder/Decoder) [Feb 22 17:22:19] == Registered file format h263, extension(s) h263 [Feb 22 17:22:19] format_h263.so => (Raw H.263 data) [Feb 22 17:22:19] == Registered file format vox, extension(s) vox [Feb 22 17:22:19] format_vox.so => (Dialogic VOX (ADPCM) File Format) [Feb 22 17:22:19] == Registered application 'BackgroundDetect' [Feb 22 17:22:19] app_talkdetect.so => (Playback with Talk Detection) [Feb 22 17:22:19] == Registered application 'SpeechCreate' [Feb 22 17:22:19] == Registered application 'SpeechLoadGrammar' [Feb 22 17:22:19] == Registered application 'SpeechUnloadGrammar' [Feb 22 17:22:19] == Registered application 'SpeechActivateGrammar' [Feb 22 17:22:19] == Registered application 'SpeechDeactivateGrammar' [Feb 22 17:22:19] == Registered application 'SpeechStart' [Feb 22 17:22:19] == Registered application 'SpeechBackground' [Feb 22 17:22:19] == Registered application 'SpeechDestroy' [Feb 22 17:22:19] == Registered application 'SpeechProcessingSound' [Feb 22 17:22:19] == Registered custom function SPEECH [Feb 22 17:22:19] == Registered custom function SPEECH_SCORE [Feb 22 17:22:19] == Registered custom function SPEECH_TEXT [Feb 22 17:22:19] == Registered custom function SPEECH_GRAMMAR [Feb 22 17:22:19] == Registered custom function SPEECH_ENGINE [Feb 22 17:22:19] app_speech_utils.so => (Dialplan Speech Applications) [Feb 22 17:22:19] == Parsing '/etc/asterisk/codecs.conf': [Feb 22 17:22:19] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/codecs.conf [Feb 22 17:22:19] Found [Feb 22 17:22:19] -- codec_adpcm: using generic PLC [Feb 22 17:22:19] WARNING[14736]: translate.c:675 __ast_register_translator: plc_samples 160 format 6 [Feb 22 17:22:19] == Registered translator 'adpcmtolin' from format adpcm to slin, cost 1 [Feb 22 17:22:19] DEBUG[14736]: translate.c:425 rebuild_matrix: Resetting translation matrix [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from g723 to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from gsm to unknown, via 3 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from gsm to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from gsm to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from gsm to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 16 cost path from gsm to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from gsm to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 6 cost path from unknown to g723, via 2 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to ulaw, via 2 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 6 cost path from unknown to unknown, via 2 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 17 cost path from unknown to unknown, via 2 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 2 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from ulaw to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from ulaw to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from ulaw to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from ulaw to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to gsm, via 3 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to gsm, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to gsm, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 4 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 4 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 4 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 4 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to gsm, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from g723 to gsm, via 3 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 7 cost path from ulaw to gsm, via 1 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from ulaw to gsm, via 3 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 7 cost path from unknown to gsm, via 1 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to gsm, via 3 [Feb 22 17:22:19] == Registered translator 'lintoadpcm' from format slin to adpcm, cost 1 [Feb 22 17:22:19] DEBUG[14736]: translate.c:425 rebuild_matrix: Resetting translation matrix [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from g723 to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from gsm to unknown, via 3 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from gsm to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from gsm to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from gsm to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from gsm to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 16 cost path from gsm to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from gsm to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 6 cost path from unknown to g723, via 2 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to ulaw, via 2 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 2 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 6 cost path from unknown to unknown, via 2 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 17 cost path from unknown to unknown, via 2 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 2 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from ulaw to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from ulaw to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from ulaw to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from ulaw to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from ulaw to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to gsm, via 3 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to gsm, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to gsm, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 4 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 4 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 4 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 4 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 4 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to gsm, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from g723 to gsm, via 3 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 7 cost path from ulaw to gsm, via 1 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from ulaw to gsm, via 3 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 7 cost path from unknown to gsm, via 1 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to gsm, via 3 [Feb 22 17:22:19] codec_adpcm.so => (Adaptive Differential PCM Coder/Decoder) [Feb 22 17:22:19] == Parsing '/etc/asterisk/oss.conf': [Feb 22 17:22:19] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/oss.conf [Feb 22 17:22:19] Found [Feb 22 17:22:19] DEBUG[14736]: channel.c:449 ast_channel_register: Registered handler for 'Console' (OSS Console Channel Driver) [Feb 22 17:22:19] == Registered channel type 'Console' (OSS Console Channel Driver) [Feb 22 17:22:19] chan_oss.so => (OSS Console Channel Driver) [Feb 22 17:22:19] == Parsing '/etc/asterisk/codecs.conf': [Feb 22 17:22:19] DEBUG[14736]: config.c:844 config_text_file_load: Parsing /etc/asterisk/codecs.conf [Feb 22 17:22:19] Found [Feb 22 17:22:19] -- codec_ulaw: using generic PLC [Feb 22 17:22:19] WARNING[14736]: translate.c:675 __ast_register_translator: plc_samples 160 format 6 [Feb 22 17:22:19] == Registered translator 'ulawtolin' from format ulaw to slin, cost 1 [Feb 22 17:22:19] DEBUG[14736]: translate.c:425 rebuild_matrix: Resetting translation matrix [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from g723 to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from gsm to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from gsm to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from gsm to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from gsm to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from gsm to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from gsm to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 2 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to ulaw, via 2 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 2 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 2 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 2 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 2 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from ulaw to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from ulaw to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from ulaw to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from ulaw to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from ulaw to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to gsm, via 3 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to gsm, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to gsm, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 4 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 4 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 4 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 4 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 4 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to gsm, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from g723 to gsm, via 3 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 7 cost path from ulaw to gsm, via 1 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from ulaw to gsm, via 3 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 7 cost path from unknown to gsm, via 1 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to gsm, via 3 [Feb 22 17:22:19] == Registered translator 'lintoulaw' from format slin to ulaw, cost 1 [Feb 22 17:22:19] DEBUG[14736]: translate.c:425 rebuild_matrix: Resetting translation matrix [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to gsm, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from g723 to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from gsm to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from gsm to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from gsm to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from gsm to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from gsm to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from gsm to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 2 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to ulaw, via 2 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 2 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 2 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 2 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 2 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from ulaw to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from ulaw to gsm, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from ulaw to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from ulaw to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from ulaw to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from ulaw to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to gsm, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to gsm, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to gsm, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to ulaw, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 4 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to gsm, via 4 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 4 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 4 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 4 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 4 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to gsm, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] DEBUG[14736]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Feb 22 17:22:19] codec_ulaw.so => (mu-Law Coder/Decoder) [Feb 22 17:22:19] == Registered custom function BLACKLIST [Feb 22 17:22:19] == Registered application 'LookupBlacklist' [Feb 22 17:22:19] app_lookupblacklist.so => (Look up Caller*ID name/number from blacklist database) [Feb 22 17:22:19] == Registered application 'GetCPEID' [Feb 22 17:22:19] app_getcpeid.so => (Get ADSI CPE ID) [Feb 22 17:22:19] pbx_realtime.so => (Realtime Switch) [Feb 22 17:22:19] == Registered file format wav, extension(s) wav [Feb 22 17:22:19] format_wav.so => (Microsoft WAV format (8000Hz Signed Linear)) [Feb 22 17:22:19] == Registered application 'ChannelRedirect' [Feb 22 17:22:19] app_channelredirect.so => (Channel Redirect) [Feb 22 17:22:19] == Registered application 'SMS' [Feb 22 17:22:19] app_sms.so => (SMS/PSTN handler) [Feb 22 17:22:19] == Registered custom function ENV [Feb 22 17:22:19] == Registered custom function STAT [Feb 22 17:22:19] func_env.so => (Environment/filesystem dialplan functions) [Feb 22 17:22:19] res_clioriginate.so => (Call origination from the CLI) [Feb 22 17:22:19] == Registered application 'Authenticate' [Feb 22 17:22:19] app_authenticate.so => (Authentication Application) [Feb 22 17:22:19] Asterisk Ready. ]1;Asterisk]2;Asterisk Console on 'unifiedpaging.messagenetsystems.com' (pid 14736)*CLI> [Feb 22 17:22:19] DEBUG[14767]: chan_sip.c:4315 find_call: = No match Their Call ID: 42fcb9a2205a226e1657915a20f1c05a@24.123.23.170 Their Tag Our tag: as5416a864 [Feb 22 17:22:19] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 000ff78d-ebb20006-6007ecd7-0ae287ec@192.168.1.165 - REGISTER (No RTP) [Feb 22 17:22:19] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:22:19] DEBUG[14753]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel IAX2/606 [Feb 22 17:22:19] DEBUG[14753]: db.c:197 ast_db_get: Unable to find key 'si-000fd3000028' in family 'iax/provisioning/cache' [Feb 22 17:22:19] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for IAX2 - 606 [Feb 22 17:22:19] DEBUG[14741]: chan_iax2.c:9600 iax2_devicestate: Checking device state for device 606 [Feb 22 17:22:19] DEBUG[14741]: chan_iax2.c:9608 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 606? addr=-1509840704, defaddr=0 maxms=0, lastms=0 [Feb 22 17:22:19] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for IAX2/606 - state 1 (Not in use) [Feb 22 17:22:19] DEBUG[14770]: app_queue.c:546 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. [Feb 22 17:22:19] DEBUG[14753]: iax2-provision.c:253 iax_provision_version: Unable to create provisioning packet for 'si-000fd3000028' [Feb 22 17:22:20] DEBUG[14767]: chan_sip.c:4315 find_call: = Found Their Call ID: 000ff78d-ebb20006-6007ecd7-0ae287ec@192.168.1.165 Their Tag Our tag: as492c69d6 [Feb 22 17:22:20] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:22:20] -- Saved useragent "CSCO/7" for peer 514 [Feb 22 17:22:20] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/514 [Feb 22 17:22:20] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 514 [Feb 22 17:22:20] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 514 [Feb 22 17:22:20] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/514 - state 1 (Not in use) [Feb 22 17:22:20] DEBUG[14771]: app_queue.c:546 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. [Feb 22 17:22:21] DEBUG[14767]: chan_sip.c:4315 find_call: = No match Their Call ID: 000ff78d-ebb20006-6007ecd7-0ae287ec@192.168.1.165 Their Tag Our tag: as492c69d6 [Feb 22 17:22:21] DEBUG[14767]: chan_sip.c:4315 find_call: = No match Their Call ID: 42fcb9a2205a226e1657915a20f1c05a@24.123.23.170 Their Tag Our tag: as5416a864 [Feb 22 17:22:21] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 55ae664767e0@24.123.23.170 - REGISTER (No RTP) [Feb 22 17:22:21] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:22:21] DEBUG[14767]: chan_sip.c:4315 find_call: = Found Their Call ID: 55ae664767e0@24.123.23.170 Their Tag 3e3880fb3b87cc4be37e8e4ec3266164 Our tag: as0e588607 [Feb 22 17:22:21] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:22:21] -- Saved useragent "Uniden SIP Phone p2 Ver BS4.77" for peer 523 [Feb 22 17:22:21] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/523 [Feb 22 17:22:21] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 523 [Feb 22 17:22:21] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 523 [Feb 22 17:22:21] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/523 - state 1 (Not in use) [Feb 22 17:22:21] DEBUG[14772]: app_queue.c:546 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. [Feb 22 17:22:21] DEBUG[14755]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel IAX2/616 [Feb 22 17:22:21] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for IAX2 - 616 [Feb 22 17:22:21] DEBUG[14741]: chan_iax2.c:9600 iax2_devicestate: Checking device state for device 616 [Feb 22 17:22:21] DEBUG[14741]: chan_iax2.c:9608 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 616? addr=-1891793590, defaddr=0 maxms=0, lastms=0 [Feb 22 17:22:21] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for IAX2/616 - state 1 (Not in use) [Feb 22 17:22:21] DEBUG[14755]: db.c:197 ast_db_get: Unable to find key 'si-000fd3000124' in family 'iax/provisioning/cache' [Feb 22 17:22:21] DEBUG[14773]: app_queue.c:546 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. [Feb 22 17:22:21] DEBUG[14755]: iax2-provision.c:253 iax_provision_version: Unable to create provisioning packet for 'si-000fd3000124' [Feb 22 17:22:23] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Feb 22 17:22:23] DEBUG[14767]: chan_sip.c:4315 find_call: = Found Their Call ID: 133d004b6870d72f25455651332ca6ca@24.123.23.170 Their Tag Our tag: as5c61520d [Feb 22 17:22:23] DEBUG[14767]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '133d004b6870d72f25455651332ca6ca@24.123.23.170' of Request 102: Match Not Found [Feb 22 17:22:23] NOTICE[14767]: chan_sip.c:12001 handle_response_peerpoke: Peer 'Broadvoice' is now Reachable. (33ms / 2000ms) [Feb 22 17:22:23] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/Broadvoice [Feb 22 17:22:23] Really destroying SIP dialog '133d004b6870d72f25455651332ca6ca@24.123.23.170' Method: OPTIONS [Feb 22 17:22:23] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - Broadvoice [Feb 22 17:22:23] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer Broadvoice [Feb 22 17:22:23] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/Broadvoice - state 1 (Not in use) [Feb 22 17:22:23] DEBUG[14774]: app_queue.c:546 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. *CLI> [Feb 22 17:22:26] DEBUG[14767]: chan_sip.c:4315 find_call: = No match Their Call ID: 55ae664767e0@24.123.23.170 Their Tag 3e3880fb3b87cc4be37e8e4ec3266164 Our tag: as0e588607 [Feb 22 17:22:26] DEBUG[14767]: chan_sip.c:4315 find_call: = No match Their Call ID: 000ff78d-ebb20006-6007ecd7-0ae287ec@192.168.1.165 Their Tag Our tag: as492c69d6 [Feb 22 17:22:26] DEBUG[14767]: chan_sip.c:4315 find_call: = No match Their Call ID: 42fcb9a2205a226e1657915a20f1c05a@24.123.23.170 Their Tag Our tag: as5416a864 [Feb 22 17:22:26] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 55ae66469f7f@24.123.23.170 - REGISTER (No RTP) [Feb 22 17:22:26] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:22:26] DEBUG[14767]: chan_sip.c:4315 find_call: = Found Their Call ID: 55ae66469f7f@24.123.23.170 Their Tag bd228682a1e81c924c359287238ef34f Our tag: as4a79073c [Feb 22 17:22:26] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:22:26] -- Saved useragent "Uniden SIP Phone p2 Ver BS4.77" for peer 528 [Feb 22 17:22:26] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/528 [Feb 22 17:22:26] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 528 [Feb 22 17:22:26] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 528 [Feb 22 17:22:26] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/528 - state 1 (Not in use) [Feb 22 17:22:26] DEBUG[14775]: app_queue.c:546 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. set debug 4[Feb 22 17:22:28] DEBUG[14767]: chan_sip.c:4315 find_call: = No match Their Call ID: 55ae66469f7f@24.123.23.170 Their Tag bd228682a1e81c924c359287238ef34f Our tag: as4a79073c [Feb 22 17:22:28] DEBUG[14767]: chan_sip.c:4315 find_call: = No match Their Call ID: 55ae664767e0@24.123.23.170 Their Tag 3e3880fb3b87cc4be37e8e4ec3266164 Our tag: as0e588607 [Feb 22 17:22:28] DEBUG[14767]: chan_sip.c:4315 find_call: = No match Their Call ID: 000ff78d-ebb20006-6007ecd7-0ae287ec@192.168.1.165 Their Tag Our tag: as492c69d6 [Feb 22 17:22:28] DEBUG[14767]: chan_sip.c:4315 find_call: = No match Their Call ID: 42fcb9a2205a226e1657915a20f1c05a@24.123.23.170 Their Tag Our tag: as5416a864 [Feb 22 17:22:28] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 55ae66476793@24.123.23.170 - REGISTER (No RTP) [Feb 22 17:22:28] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:22:28] DEBUG[14767]: chan_sip.c:4315 find_call: = Found Their Call ID: 55ae66476793@24.123.23.170 Their Tag bc48b99ece94334fc34c5a38f540247e Our tag: as3e370f2c [Feb 22 17:22:28] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:22:28] -- Saved useragent "Uniden SIP Phone p2 Ver BS4.77" for peer 525 [Feb 22 17:22:28] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/525 [Feb 22 17:22:28] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 525 [Feb 22 17:22:28] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 525 [Feb 22 17:22:28] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/525 - state 1 (Not in use) [Feb 22 17:22:28] DEBUG[14776]: app_queue.c:546 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. Core debug was 6 and is now 4 The 'set debug' command is deprecated and will be removed in a future release. Please use 'core set debug' instead. *CLI> core set debug 4 Core debug is at least 4 *CLI> core ser t ver[Feb 22 17:22:40] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 55ae66469f11@24.123.23.170 - REGISTER (No RTP) [Feb 22 17:22:40] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER b[Feb 22 17:22:40] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:22:40] -- Saved useragent "Uniden SIP Phone p2 Ver BS4.77" for peer 522 [Feb 22 17:22:40] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/522 [Feb 22 17:22:40] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 522 [Feb 22 17:22:40] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 522 [Feb 22 17:22:40] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/522 - state 1 (Not in use) [Feb 22 17:22:40] DEBUG[14780]: app_queue.c:546 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. ose 4 Verbosity was 7 and is now 4 *CLI> [Feb 22 17:22:42] DEBUG[14760]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel IAX2/651 [Feb 22 17:22:42] DEBUG[14760]: db.c:197 ast_db_get: Unable to find key 'si-000364000738' in family 'iax/provisioning/cache' [Feb 22 17:22:42] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for IAX2 - 651 [Feb 22 17:22:42] DEBUG[14741]: chan_iax2.c:9600 iax2_devicestate: Checking device state for device 651 [Feb 22 17:22:42] DEBUG[14741]: chan_iax2.c:9608 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 651? addr=-1459509056, defaddr=0 maxms=0, lastms=0 [Feb 22 17:22:42] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for IAX2/651 - state 1 (Not in use) [Feb 22 17:22:42] DEBUG[14781]: app_queue.c:546 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. [Feb 22 17:22:42] DEBUG[14760]: iax2-provision.c:253 iax_provision_version: Unable to create provisioning packet for 'si-000364000738' [Feb 22 17:22:43] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 42fcb9a2205a226e1657915a20f1c05a@24.123.23.170 - REGISTER (No RTP) [Feb 22 17:22:43] DEBUG[14767]: chan_sip.c:7214 transmit_register: Scheduled a registration timeout for sip.broadvoice.com id #102 [Feb 22 17:22:43] REGISTER attempt 1 to 3173241052@sip.broadvoice.com [Feb 22 17:22:43] DEBUG[14767]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '42fcb9a2205a226e1657915a20f1c05a@24.123.23.170' of Request 103: Match Not Found [Feb 22 17:22:43] DEBUG[14767]: chan_sip.c:11906 handle_response_register: Registration successful [Feb 22 17:22:43] DEBUG[14767]: chan_sip.c:11909 handle_response_register: Cancelling timeout 102 sip debug[Feb 22 17:22:45] DEBUG[14764]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel IAX2/614 [Feb 22 17:22:45] DEBUG[14764]: db.c:197 ast_db_get: Unable to find key 'si-000fd300002e' in family 'iax/provisioning/cache' [Feb 22 17:22:45] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for IAX2 - 614 [Feb 22 17:22:45] DEBUG[14741]: chan_iax2.c:9600 iax2_devicestate: Checking device state for device 614 [Feb 22 17:22:45] DEBUG[14741]: chan_iax2.c:9608 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 614? addr=-1493063488, defaddr=0 maxms=0, lastms=0 [Feb 22 17:22:45] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for IAX2/614 - state 1 (Not in use) [Feb 22 17:22:45] DEBUG[14782]: app_queue.c:546 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. [Feb 22 17:22:45] DEBUG[14764]: iax2-provision.c:253 iax_provision_version: Unable to create provisioning packet for 'si-000fd300002e' SIP Debugging enabled The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead. *CLI> [Feb 22 17:22:48] <--- SIP read from 68.58.36.157:5060 ---> <-------------> [Feb 22 17:22:48] --- (0 headers 0 lines) Nat keepalive --- [Feb 22 17:22:48] DEBUG[14786]: manager.c:1931 process_message: Manager received command 'Login' [Feb 22 17:22:48] == Parsing '/etc/asterisk/manager.conf': [Feb 22 17:22:48] DEBUG[14786]: config.c:844 config_text_file_load: Parsing /etc/asterisk/manager.conf [Feb 22 17:22:48] Found [Feb 22 17:22:48] DEBUG[14786]: acl.c:199 ast_append_ha: 127.0.0.1/255.255.255.0/255.255.255.0 appended to acl for peer [Feb 22 17:22:48] DEBUG[14786]: acl.c:213 ast_apply_ha: ##### Testing 127.0.0.1 with 127.0.0.0 [Feb 22 17:22:48] == Manager 'MessageNet' logged on from 127.0.0.1 [Feb 22 17:22:48] DEBUG[14786]: manager.c:1931 process_message: Manager received command 'Command' [Feb 22 17:22:48] DEBUG[14786]: manager.c:1931 process_message: Manager received command 'Command' [Feb 22 17:22:48] DEBUG[14786]: manager.c:1931 process_message: Manager received command 'Logoff' [Feb 22 17:22:48] == Manager 'MessageNet' logged off from 127.0.0.1 si pset deb[Feb 22 17:22:51] DEBUG[14767]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '42fcb9a2205a226e1657915a20f1c05a@24.123.23.170' [Feb 22 17:22:51] DEBUG[14767]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 42fcb9a2205a226e1657915a20f1c05a@24.123.23.170 [Feb 22 17:22:51] Really destroying SIP dialog '42fcb9a2205a226e1657915a20f1c05a@24.123.23.170' Method: REGISTER ug[Feb 22 17:22:52] DEBUG[14767]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '000ff78d-ebb20006-6007ecd7-0ae287ec@192.168.1.165' [Feb 22 17:22:52] DEBUG[14767]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 000ff78d-ebb20006-6007ecd7-0ae287ec@192.168.1.165 [Feb 22 17:22:52] Really destroying SIP dialog '000ff78d-ebb20006-6007ecd7-0ae287ec@192.168.1.165' Method: REGISTER No such command 'si pset' (type 'help' for help) *CLI> [Feb 22 17:22:52] <--- 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=z9hG4bKacc56924fb6523422644ad6a4103042c9 Call-ID: 55ae6647666e@24.123.23.170 CSeq: 64472 REGISTER From: ;tag=7b3491472edaac4474dd940c71fce11d To: Contact: ;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="735eb1c6", algorithm=MD5, uri="sip:24.123.23.170", username="526", response="470237621dac03839035cfb50c5ff6e4" <-------------> [Feb 22 17:22:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:22:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bKacc56924fb6523422644ad6a4103042c9 (82) [Feb 22 17:22:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: Call-ID: 55ae6647666e@24.123.23.170 (35) [Feb 22 17:22:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: CSeq: 64472 REGISTER (20) [Feb 22 17:22:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: ;tag=7b3491472edaac4474dd940c71fce11d (66) [Feb 22 17:22:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: To: (27) [Feb 22 17:22:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Feb 22 17:22:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:22:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 22 17:22:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:22:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:22:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:22:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="735eb1c6", algorithm=MD5, uri="sip:24.123.23.170", username="526", response="470237621dac03839035cfb50c5ff6e4" (167) [Feb 22 17:22:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 13: (0) [Feb 22 17:22:52] --- (13 headers 0 lines) --- [Feb 22 17:22:52] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 55ae6647666e@24.123.23.170 - REGISTER (No RTP) [Feb 22 17:22:52] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:22:52] Using latest REGISTER request as basis request [Feb 22 17:22:52] Sending to 192.168.1.66 : 5060 (no NAT) [Feb 22 17:22:52] <--- 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=z9hG4bKacc56924fb6523422644ad6a4103042c9;received=192.168.1.66 From: ;tag=7b3491472edaac4474dd940c71fce11d To: Call-ID: 55ae6647666e@24.123.23.170 CSeq: 64472 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:22:52] <--- 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=z9hG4bKacc56924fb6523422644ad6a4103042c9;received=192.168.1.66 From: ;tag=7b3491472edaac4474dd940c71fce11d To: ;tag=as1e814949 Call-ID: 55ae6647666e@24.123.23.170 CSeq: 64472 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="39b1f81b" Content-Length: 0 <------------> [Feb 22 17:22:52] Scheduling destruction of SIP dialog '55ae6647666e@24.123.23.170' in 32000 ms (Method: REGISTER) [Feb 22 17:22:52] <--- 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=z9hG4bKe62c1be5f636d2d4a0da7866ef0121188 CSeq: 64473 REGISTER Call-ID: 55ae6647666e@24.123.23.170 From: ;tag=7b3491472edaac4474dd940c71fce11d To: Contact: ;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="39b1f81b", algorithm=MD5, uri="sip:24.123.23.170", username="526", response="5469d046dc161dfca54960cb9bad49ed" <-------------> [Feb 22 17:22:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:22:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bKe62c1be5f636d2d4a0da7866ef0121188 (82) [Feb 22 17:22:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: CSeq: 64473 REGISTER (20) [Feb 22 17:22:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: Call-ID: 55ae6647666e@24.123.23.170 (35) [Feb 22 17:22:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: ;tag=7b3491472edaac4474dd940c71fce11d (66) [Feb 22 17:22:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: To: (27) [Feb 22 17:22:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Feb 22 17:22:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:22:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 22 17:22:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:22:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:22:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:22:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="39b1f81b", algorithm=MD5, uri="sip:24.123.23.170", username="526", response="5469d046dc161dfca54960cb9bad49ed" (167) [Feb 22 17:22:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 13: (0) [Feb 22 17:22:52] --- (13 headers 0 lines) --- [Feb 22 17:22:52] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:22:52] Using latest REGISTER request as basis request [Feb 22 17:22:52] Sending to 192.168.1.66 : 5060 (no NAT) [Feb 22 17:22:52] <--- 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=z9hG4bKe62c1be5f636d2d4a0da7866ef0121188;received=192.168.1.66 From: ;tag=7b3491472edaac4474dd940c71fce11d To: Call-ID: 55ae6647666e@24.123.23.170 CSeq: 64473 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:22:52] -- Saved useragent "Uniden SIP Phone p2 Ver BS4.77" for peer 526 [Feb 22 17:22:52] <--- 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=z9hG4bKe62c1be5f636d2d4a0da7866ef0121188;received=192.168.1.66 From: ;tag=7b3491472edaac4474dd940c71fce11d To: ;tag=as1e814949 Call-ID: 55ae6647666e@24.123.23.170 CSeq: 64473 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Thu, 22 Feb 2007 22:22:52 GMT Content-Length: 0 <------------> [Feb 22 17:22:52] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/526 [Feb 22 17:22:52] Scheduling destruction of SIP dialog '55ae6647666e@24.123.23.170' in 32000 ms (Method: REGISTER) [Feb 22 17:22:52] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 526 [Feb 22 17:22:52] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 526 [Feb 22 17:22:52] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/526 - state 1 (Not in use) [Feb 22 17:22:52] DEBUG[14787]: app_queue.c:546 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. [Feb 22 17:22:53] DEBUG[14767]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '55ae664767e0@24.123.23.170' [Feb 22 17:22:53] DEBUG[14767]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 55ae664767e0@24.123.23.170 [Feb 22 17:22:53] Really destroying SIP dialog '55ae664767e0@24.123.23.170' Method: REGISTER [Feb 22 17:22:54] <--- 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=z9hG4bK4e5183f1 From: To: Call-ID: 000ff78d-ebb20007-74946e6d-3266b768@192.168.1.165 Date: Thu, 22 Feb 2007 22:24:27 GMT CSeq: 49724 REGISTER User-Agent: CSCO/7 Contact: Content-Length: 0 Expires: 60 <-------------> [Feb 22 17:22:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:22:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK4e5183f1 (58) [Feb 22 17:22:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: (40) [Feb 22 17:22:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (38) [Feb 22 17:22:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 000ff78d-ebb20007-74946e6d-3266b768@192.168.1.165 (58) [Feb 22 17:22:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Date: Thu, 22 Feb 2007 22:24:27 GMT (35) [Feb 22 17:22:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: CSeq: 49724 REGISTER (20) [Feb 22 17:22:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: CSCO/7 (18) [Feb 22 17:22:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Contact: (37) [Feb 22 17:22:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:22:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Expires: 60 (11) [Feb 22 17:22:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: (0) [Feb 22 17:22:54] --- (11 headers 0 lines) --- [Feb 22 17:22:54] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 000ff78d-ebb20007-74946e6d-3266b768@192.168.1.165 - REGISTER (No RTP) [Feb 22 17:22:54] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:22:54] Using latest REGISTER request as basis request [Feb 22 17:22:54] Sending to 192.168.1.165 : 5060 (no NAT) [Feb 22 17:22:54] <--- 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=z9hG4bK4e5183f1;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20007-74946e6d-3266b768@192.168.1.165 CSeq: 49724 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:22:54] <--- 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=z9hG4bK4e5183f1;received=192.168.1.165 From: To: ;tag=as6c95ad9c Call-ID: 000ff78d-ebb20007-74946e6d-3266b768@192.168.1.165 CSeq: 49724 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="618e7fd0" Content-Length: 0 <------------> [Feb 22 17:22:54] Scheduling destruction of SIP dialog '000ff78d-ebb20007-74946e6d-3266b768@192.168.1.165' in 32000 ms (Method: REGISTER) [Feb 22 17:22:54] <--- 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=z9hG4bK0a4d06da From: To: Call-ID: 000ff78d-ebb20007-74946e6d-3266b768@192.168.1.165 Date: Thu, 22 Feb 2007 22:24:28 GMT CSeq: 49725 REGISTER User-Agent: CSCO/7 Contact: Authorization: Digest username="515",realm="asterisk",uri="sip:24.123.23.170",response="583d0e44910df74afaaaa00c27cfd209",nonce="618e7fd0",algorithm=MD5 Content-Length: 0 Expires: 60 <-------------> [Feb 22 17:22:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:22:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK0a4d06da (58) [Feb 22 17:22:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: (40) [Feb 22 17:22:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (38) [Feb 22 17:22:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 000ff78d-ebb20007-74946e6d-3266b768@192.168.1.165 (58) [Feb 22 17:22:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Date: Thu, 22 Feb 2007 22:24:28 GMT (35) [Feb 22 17:22:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: CSeq: 49725 REGISTER (20) [Feb 22 17:22:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: CSCO/7 (18) [Feb 22 17:22:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Contact: (37) [Feb 22 17:22:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Authorization: Digest username="515",realm="asterisk",uri="sip:24.123.23.170",response="583d0e44910df74afaaaa00c27cfd209",nonce="618e7fd0",algorithm=MD5 (152) [Feb 22 17:22:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Content-Length: 0 (17) [Feb 22 17:22:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Expires: 60 (11) [Feb 22 17:22:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: (0) [Feb 22 17:22:54] --- (12 headers 0 lines) --- [Feb 22 17:22:54] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:22:54] Using latest REGISTER request as basis request [Feb 22 17:22:54] Sending to 192.168.1.165 : 5060 (no NAT) [Feb 22 17:22:54] <--- 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=z9hG4bK0a4d06da;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20007-74946e6d-3266b768@192.168.1.165 CSeq: 49725 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:22:54] -- Saved useragent "CSCO/7" for peer 515 [Feb 22 17:22:54] <--- 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=z9hG4bK0a4d06da;received=192.168.1.165 From: To: ;tag=as6c95ad9c Call-ID: 000ff78d-ebb20007-74946e6d-3266b768@192.168.1.165 CSeq: 49725 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Thu, 22 Feb 2007 22:22:54 GMT Content-Length: 0 <------------> [Feb 22 17:22:54] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/515 [Feb 22 17:22:54] Scheduling destruction of SIP dialog '000ff78d-ebb20007-74946e6d-3266b768@192.168.1.165' in 32000 ms (Method: REGISTER) [Feb 22 17:22:54] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 515 [Feb 22 17:22:54] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 515 [Feb 22 17:22:54] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/515 - state 1 (Not in use) [Feb 22 17:22:54] DEBUG[14788]: app_queue.c:546 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. [Feb 22 17:22:55] <--- 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=z9hG4bKed8f30c894bafdba77ddf98b5a8a21bb0 Call-ID: 55ae664696c0@24.123.23.170 CSeq: 58869 REGISTER From: ;tag=f1615faaf03b33b3470c7f91a0b47a6d To: Contact: ;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="5407d599", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="ce70435e30e7977a67b8f005ea0ccf62" <-------------> [Feb 22 17:22:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:22:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.98:5060;branch=z9hG4bKed8f30c894bafdba77ddf98b5a8a21bb0 (82) [Feb 22 17:22:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: Call-ID: 55ae664696c0@24.123.23.170 (35) [Feb 22 17:22:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: CSeq: 58869 REGISTER (20) [Feb 22 17:22:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: ;tag=f1615faaf03b33b3470c7f91a0b47a6d (66) [Feb 22 17:22:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: To: (27) [Feb 22 17:22:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Feb 22 17:22:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:22:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 22 17:22:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:22:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:22:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:22:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="5407d599", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="ce70435e30e7977a67b8f005ea0ccf62" (167) [Feb 22 17:22:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 13: (0) [Feb 22 17:22:55] --- (13 headers 0 lines) --- [Feb 22 17:22:55] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 55ae664696c0@24.123.23.170 - REGISTER (No RTP) [Feb 22 17:22:55] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:22:55] Using latest REGISTER request as basis request [Feb 22 17:22:55] Sending to 192.168.1.98 : 5060 (no NAT) [Feb 22 17:22:55] <--- 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=z9hG4bKed8f30c894bafdba77ddf98b5a8a21bb0;received=192.168.1.98 From: ;tag=f1615faaf03b33b3470c7f91a0b47a6d To: Call-ID: 55ae664696c0@24.123.23.170 CSeq: 58869 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:22:55] <--- 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=z9hG4bKed8f30c894bafdba77ddf98b5a8a21bb0;received=192.168.1.98 From: ;tag=f1615faaf03b33b3470c7f91a0b47a6d To: ;tag=as504aeea5 Call-ID: 55ae664696c0@24.123.23.170 CSeq: 58869 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="41d99ab7" Content-Length: 0 <------------> [Feb 22 17:22:55] Scheduling destruction of SIP dialog '55ae664696c0@24.123.23.170' in 32000 ms (Method: REGISTER) [Feb 22 17:22:55] <--- 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=z9hG4bKi80d5a6e875b344f00c71aa7d2cadb637 CSeq: 58870 REGISTER Call-ID: 55ae664696c0@24.123.23.170 From: ;tag=f1615faaf03b33b3470c7f91a0b47a6d To: Contact: ;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="41d99ab7", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="641202dafb78240bec6b7287e55fc72a" <-------------> [Feb 22 17:22:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:22:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.98:5060;branch=z9hG4bKi80d5a6e875b344f00c71aa7d2cadb637 (82) [Feb 22 17:22:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: CSeq: 58870 REGISTER (20) [Feb 22 17:22:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: Call-ID: 55ae664696c0@24.123.23.170 (35) [Feb 22 17:22:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: ;tag=f1615faaf03b33b3470c7f91a0b47a6d (66) [Feb 22 17:22:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: To: (27) [Feb 22 17:22:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Feb 22 17:22:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:22:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 22 17:22:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:22:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:22:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:22:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="41d99ab7", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="641202dafb78240bec6b7287e55fc72a" (167) [Feb 22 17:22:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 13: (0) [Feb 22 17:22:55] --- (13 headers 0 lines) --- [Feb 22 17:22:55] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:22:55] Using latest REGISTER request as basis request [Feb 22 17:22:55] Sending to 192.168.1.98 : 5060 (no NAT) [Feb 22 17:22:55] <--- 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=z9hG4bKi80d5a6e875b344f00c71aa7d2cadb637;received=192.168.1.98 From: ;tag=f1615faaf03b33b3470c7f91a0b47a6d To: Call-ID: 55ae664696c0@24.123.23.170 CSeq: 58870 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:22:55] -- Saved useragent "Uniden SIP Phone p2 Ver BS4.77" for peer 529 [Feb 22 17:22:55] <--- 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=z9hG4bKi80d5a6e875b344f00c71aa7d2cadb637;received=192.168.1.98 From: ;tag=f1615faaf03b33b3470c7f91a0b47a6d To: ;tag=as504aeea5 Call-ID: 55ae664696c0@24.123.23.170 CSeq: 58870 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Thu, 22 Feb 2007 22:22:55 GMT Content-Length: 0 <------------> [Feb 22 17:22:55] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/529 [Feb 22 17:22:55] Scheduling destruction of SIP dialog '55ae664696c0@24.123.23.170' in 32000 ms (Method: REGISTER) [Feb 22 17:22:55] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 529 [Feb 22 17:22:55] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 529 [Feb 22 17:22:55] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/529 - state 1 (Not in use) [Feb 22 17:22:55] DEBUG[14789]: app_queue.c:546 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. [Feb 22 17:22:56] <--- SIP read from 192.168.1.86:5062 ---> <-------------> [Feb 22 17:22:56] --- (0 headers 0 lines) Nat keepalive --- [Feb 22 17:22:57] DEBUG[14755]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel IAX2/606 [Feb 22 17:22:57] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for IAX2 - 606 [Feb 22 17:22:57] DEBUG[14741]: chan_iax2.c:9600 iax2_devicestate: Checking device state for device 606 [Feb 22 17:22:57] DEBUG[14741]: chan_iax2.c:9608 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 606? addr=-1509840704, defaddr=0 maxms=0, lastms=0 [Feb 22 17:22:57] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for IAX2/606 - state 1 (Not in use) [Feb 22 17:22:57] DEBUG[14790]: app_queue.c:546 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. [Feb 22 17:22:58] <--- 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=z9hG4bK45638458 From: To: Call-ID: 000ff78d-ebb20003-42dc795d-3e449dd6@192.168.1.165 Date: Thu, 22 Feb 2007 22:24:32 GMT CSeq: 49703 REGISTER User-Agent: CSCO/7 Contact: Content-Length: 0 Expires: 60 <-------------> [Feb 22 17:22:58] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:22:58] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK45638458 (58) [Feb 22 17:22:58] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: (40) [Feb 22 17:22:58] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (38) [Feb 22 17:22:58] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 000ff78d-ebb20003-42dc795d-3e449dd6@192.168.1.165 (58) [Feb 22 17:22:58] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Date: Thu, 22 Feb 2007 22:24:32 GMT (35) [Feb 22 17:22:58] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: CSeq: 49703 REGISTER (20) [Feb 22 17:22:58] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: CSCO/7 (18) [Feb 22 17:22:58] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Contact: (37) [Feb 22 17:22:58] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:22:58] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Expires: 60 (11) [Feb 22 17:22:58] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: (0) [Feb 22 17:22:58] --- (11 headers 0 lines) --- [Feb 22 17:22:58] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 000ff78d-ebb20003-42dc795d-3e449dd6@192.168.1.165 - REGISTER (No RTP) [Feb 22 17:22:58] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:22:58] Using latest REGISTER request as basis request [Feb 22 17:22:58] Sending to 192.168.1.165 : 5060 (no NAT) [Feb 22 17:22:58] <--- 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=z9hG4bK45638458;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20003-42dc795d-3e449dd6@192.168.1.165 CSeq: 49703 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:22:58] <--- 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=z9hG4bK45638458;received=192.168.1.165 From: To: ;tag=as47df7ef1 Call-ID: 000ff78d-ebb20003-42dc795d-3e449dd6@192.168.1.165 CSeq: 49703 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="58744882" Content-Length: 0 <------------> [Feb 22 17:22:58] Scheduling destruction of SIP dialog '000ff78d-ebb20003-42dc795d-3e449dd6@192.168.1.165' in 32000 ms (Method: REGISTER) [Feb 22 17:22:58] DEBUG[14767]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '55ae66469f7f@24.123.23.170' [Feb 22 17:22:58] DEBUG[14767]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 55ae66469f7f@24.123.23.170 [Feb 22 17:22:58] Really destroying SIP dialog '55ae66469f7f@24.123.23.170' Method: REGISTER [Feb 22 17:22:58] <--- 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=z9hG4bK6aa84930 From: To: Call-ID: 000ff78d-ebb20003-42dc795d-3e449dd6@192.168.1.165 Date: Thu, 22 Feb 2007 22:24:32 GMT CSeq: 49704 REGISTER User-Agent: CSCO/7 Contact: Authorization: Digest username="511",realm="asterisk",uri="sip:24.123.23.170",response="8a7582f30ed0ca6618ad5a9e36cf83df",nonce="58744882",algorithm=MD5 Content-Length: 0 Expires: 60 <-------------> [Feb 22 17:22:58] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:22:58] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK6aa84930 (58) [Feb 22 17:22:58] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: (40) [Feb 22 17:22:58] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (38) [Feb 22 17:22:58] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 000ff78d-ebb20003-42dc795d-3e449dd6@192.168.1.165 (58) [Feb 22 17:22:58] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Date: Thu, 22 Feb 2007 22:24:32 GMT (35) [Feb 22 17:22:58] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: CSeq: 49704 REGISTER (20) [Feb 22 17:22:58] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: CSCO/7 (18) [Feb 22 17:22:58] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Contact: (37) [Feb 22 17:22:58] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Authorization: Digest username="511",realm="asterisk",uri="sip:24.123.23.170",response="8a7582f30ed0ca6618ad5a9e36cf83df",nonce="58744882",algorithm=MD5 (152) [Feb 22 17:22:58] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Content-Length: 0 (17) [Feb 22 17:22:58] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Expires: 60 (11) [Feb 22 17:22:58] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: (0) [Feb 22 17:22:58] --- (12 headers 0 lines) --- [Feb 22 17:22:58] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:22:58] Using latest REGISTER request as basis request [Feb 22 17:22:58] Sending to 192.168.1.165 : 5060 (no NAT) [Feb 22 17:22:58] <--- 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=z9hG4bK6aa84930;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20003-42dc795d-3e449dd6@192.168.1.165 CSeq: 49704 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:22:58] -- Saved useragent "CSCO/7" for peer 511 [Feb 22 17:22:58] <--- 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=z9hG4bK6aa84930;received=192.168.1.165 From: To: ;tag=as47df7ef1 Call-ID: 000ff78d-ebb20003-42dc795d-3e449dd6@192.168.1.165 CSeq: 49704 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Thu, 22 Feb 2007 22:22:58 GMT Content-Length: 0 <------------> [Feb 22 17:22:58] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/511 [Feb 22 17:22:58] Scheduling destruction of SIP dialog '000ff78d-ebb20003-42dc795d-3e449dd6@192.168.1.165' in 32000 ms (Method: REGISTER) [Feb 22 17:22:58] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 511 [Feb 22 17:22:58] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 511 [Feb 22 17:22:58] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/511 - state 1 (Not in use) [Feb 22 17:22:58] DEBUG[14791]: app_queue.c:546 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. [Feb 22 17:23:00] DEBUG[14759]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel IAX2/616 [Feb 22 17:23:00] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for IAX2 - 616 [Feb 22 17:23:00] DEBUG[14741]: chan_iax2.c:9600 iax2_devicestate: Checking device state for device 616 [Feb 22 17:23:00] DEBUG[14741]: chan_iax2.c:9608 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 616? addr=-1891793590, defaddr=0 maxms=0, lastms=0 [Feb 22 17:23:00] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for IAX2/616 - state 1 (Not in use) [Feb 22 17:23:00] DEBUG[14792]: app_queue.c:546 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. [Feb 22 17:23:00] DEBUG[14767]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '55ae66476793@24.123.23.170' [Feb 22 17:23:00] DEBUG[14767]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 55ae66476793@24.123.23.170 [Feb 22 17:23:00] Really destroying SIP dialog '55ae66476793@24.123.23.170' Method: REGISTER [Feb 22 17:23:03] <--- 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=z9hG4bK08e2aa89 From: To: Call-ID: 000ff78d-ebb20005-4fe621ee-067da175@192.168.1.165 Date: Thu, 22 Feb 2007 22:24:36 GMT CSeq: 49745 REGISTER User-Agent: CSCO/7 Contact: Content-Length: 0 Expires: 60 <-------------> [Feb 22 17:23:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:23:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK08e2aa89 (58) [Feb 22 17:23:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: (40) [Feb 22 17:23:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (38) [Feb 22 17:23:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 000ff78d-ebb20005-4fe621ee-067da175@192.168.1.165 (58) [Feb 22 17:23:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Date: Thu, 22 Feb 2007 22:24:36 GMT (35) [Feb 22 17:23:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: CSeq: 49745 REGISTER (20) [Feb 22 17:23:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: CSCO/7 (18) [Feb 22 17:23:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Contact: (37) [Feb 22 17:23:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:23:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Expires: 60 (11) [Feb 22 17:23:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: (0) [Feb 22 17:23:03] --- (11 headers 0 lines) --- [Feb 22 17:23:03] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 000ff78d-ebb20005-4fe621ee-067da175@192.168.1.165 - REGISTER (No RTP) [Feb 22 17:23:03] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:23:03] Using latest REGISTER request as basis request [Feb 22 17:23:03] Sending to 192.168.1.165 : 5060 (no NAT) [Feb 22 17:23:03] <--- 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=z9hG4bK08e2aa89;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20005-4fe621ee-067da175@192.168.1.165 CSeq: 49745 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:23:03] <--- 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=z9hG4bK08e2aa89;received=192.168.1.165 From: To: ;tag=as00bcb2ae Call-ID: 000ff78d-ebb20005-4fe621ee-067da175@192.168.1.165 CSeq: 49745 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3719bfb1" Content-Length: 0 <------------> [Feb 22 17:23:03] Scheduling destruction of SIP dialog '000ff78d-ebb20005-4fe621ee-067da175@192.168.1.165' in 32000 ms (Method: REGISTER) [Feb 22 17:23:03] <--- 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=z9hG4bK0adc93a2 From: To: Call-ID: 000ff78d-ebb20005-4fe621ee-067da175@192.168.1.165 Date: Thu, 22 Feb 2007 22:24:37 GMT CSeq: 49746 REGISTER User-Agent: CSCO/7 Contact: Authorization: Digest username="513",realm="asterisk",uri="sip:24.123.23.170",response="72031134e424630103397cfedad8fcc4",nonce="3719bfb1",algorithm=MD5 Content-Length: 0 Expires: 60 <-------------> [Feb 22 17:23:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:23:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK0adc93a2 (58) [Feb 22 17:23:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: (40) [Feb 22 17:23:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (38) [Feb 22 17:23:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 000ff78d-ebb20005-4fe621ee-067da175@192.168.1.165 (58) [Feb 22 17:23:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Date: Thu, 22 Feb 2007 22:24:37 GMT (35) [Feb 22 17:23:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: CSeq: 49746 REGISTER (20) [Feb 22 17:23:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: CSCO/7 (18) [Feb 22 17:23:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Contact: (37) [Feb 22 17:23:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Authorization: Digest username="513",realm="asterisk",uri="sip:24.123.23.170",response="72031134e424630103397cfedad8fcc4",nonce="3719bfb1",algorithm=MD5 (152) [Feb 22 17:23:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Content-Length: 0 (17) [Feb 22 17:23:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Expires: 60 (11) [Feb 22 17:23:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: (0) [Feb 22 17:23:03] --- (12 headers 0 lines) --- [Feb 22 17:23:03] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:23:03] Using latest REGISTER request as basis request [Feb 22 17:23:03] Sending to 192.168.1.165 : 5060 (no NAT) [Feb 22 17:23:03] <--- 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=z9hG4bK0adc93a2;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20005-4fe621ee-067da175@192.168.1.165 CSeq: 49746 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:23:03] -- Saved useragent "CSCO/7" for peer 513 [Feb 22 17:23:03] <--- 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=z9hG4bK0adc93a2;received=192.168.1.165 From: To: ;tag=as00bcb2ae Call-ID: 000ff78d-ebb20005-4fe621ee-067da175@192.168.1.165 CSeq: 49746 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Thu, 22 Feb 2007 22:23:03 GMT Content-Length: 0 <------------> [Feb 22 17:23:03] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/513 [Feb 22 17:23:03] Scheduling destruction of SIP dialog '000ff78d-ebb20005-4fe621ee-067da175@192.168.1.165' in 32000 ms (Method: REGISTER) [Feb 22 17:23:03] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 513 [Feb 22 17:23:03] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 513 [Feb 22 17:23:03] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/513 - state 1 (Not in use) [Feb 22 17:23:03] DEBUG[14793]: app_queue.c:546 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. [Feb 22 17:23:06] <--- SIP read from 192.168.1.86:5060 ---> <-------------> [Feb 22 17:23:06] --- (0 headers 0 lines) Nat keepalive --- [Feb 22 17:23:07] NOTICE[14767]: chan_sip.c:7055 sip_reregister: -- Re-registration for 3173241052@sip.broadvoice.com [Feb 22 17:23:07] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 42fcb9a2205a226e1657915a20f1c05a@24.123.23.170 - REGISTER (No RTP) [Feb 22 17:23:07] DEBUG[14767]: chan_sip.c:7214 transmit_register: Scheduled a registration timeout for sip.broadvoice.com id #121 [Feb 22 17:23:07] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:sip.broadvoice.com SIP/2.0 (39) [Feb 22 17:23:07] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK4b51d317;rport (64) [Feb 22 17:23:07] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: ;tag=as5d88a6ff (56) [Feb 22 17:23:07] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (39) [Feb 22 17:23:07] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 42fcb9a2205a226e1657915a20f1c05a@24.123.23.170 (55) [Feb 22 17:23:07] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: CSeq: 104 REGISTER (18) [Feb 22 17:23:07] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Feb 22 17:23:07] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: Max-Forwards: 70 (16) [Feb 22 17:23:07] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Expires: 120 (12) [Feb 22 17:23:07] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Contact: (39) [Feb 22 17:23:07] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Event: registration (19) [Feb 22 17:23:07] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Content-Length: 0 (17) [Feb 22 17:23:07] REGISTER 12 headers, 0 lines [Feb 22 17:23:07] REGISTER attempt 1 to 3173241052@sip.broadvoice.com [Feb 22 17:23:07] 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=z9hG4bK4b51d317;rport From: ;tag=as5d88a6ff To: Call-ID: 42fcb9a2205a226e1657915a20f1c05a@24.123.23.170 CSeq: 104 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: Event: registration Content-Length: 0 --- [Feb 22 17:23:07] DEBUG[14767]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #122 [Feb 22 17:23:07] <--- SIP read from 147.135.12.128:5060 ---> SIP/2.0 200 OK Call-ID: 42fcb9a2205a226e1657915a20f1c05a@24.123.23.170 CSeq: 104 REGISTER From: ;tag=as5d88a6ff To: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK4b51d317 Contact: Expires: 30 Event: registration Content-Length: 0 <-------------> [Feb 22 17:23:07] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 22 17:23:07] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Call-ID: 42fcb9a2205a226e1657915a20f1c05a@24.123.23.170 (55) [Feb 22 17:23:07] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: CSeq: 104 REGISTER (18) [Feb 22 17:23:07] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: From: ;tag=as5d88a6ff (56) [Feb 22 17:23:07] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: To: (39) [Feb 22 17:23:07] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK4b51d317 (58) [Feb 22 17:23:07] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: (39) [Feb 22 17:23:07] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: Expires: 30 (11) [Feb 22 17:23:07] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Event: registration (19) [Feb 22 17:23:07] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (20) [Feb 22 17:23:07] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: (0) [Feb 22 17:23:07] --- (10 headers 0 lines) --- [Feb 22 17:23:07] DEBUG[14767]: chan_sip.c:2061 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #122 [Feb 22 17:23:07] DEBUG[14767]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '42fcb9a2205a226e1657915a20f1c05a@24.123.23.170' of Request 104: Match Not Found [Feb 22 17:23:07] DEBUG[14767]: chan_sip.c:11906 handle_response_register: Registration successful [Feb 22 17:23:07] DEBUG[14767]: chan_sip.c:11909 handle_response_register: Cancelling timeout 121 [Feb 22 17:23:07] Scheduling destruction of SIP dialog '42fcb9a2205a226e1657915a20f1c05a@24.123.23.170' in 32000 ms (Method: REGISTER) [Feb 22 17:23:07] NOTICE[14767]: chan_sip.c:11961 handle_response_register: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 23 s) [Feb 22 17:23:08] <--- SIP read from 68.58.36.157:5060 ---> <-------------> [Feb 22 17:23:08] --- (0 headers 0 lines) Nat keepalive --- [Feb 22 17:23:12] DEBUG[14767]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '55ae66469f11@24.123.23.170' [Feb 22 17:23:12] DEBUG[14767]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 55ae66469f11@24.123.23.170 [Feb 22 17:23:12] Really destroying SIP dialog '55ae66469f11@24.123.23.170' Method: REGISTER [Feb 22 17:23:15] DEBUG[14767]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '42fcb9a2205a226e1657915a20f1c05a@24.123.23.170' [Feb 22 17:23:15] DEBUG[14767]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 42fcb9a2205a226e1657915a20f1c05a@24.123.23.170 [Feb 22 17:23:15] Really destroying SIP dialog '42fcb9a2205a226e1657915a20f1c05a@24.123.23.170' Method: REGISTER [Feb 22 17:23:15] <--- 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=z9hG4bK54a8e638 From: To: Call-ID: 000ff78d-ebb20002-237ee55f-6d451a9c@192.168.1.165 Date: Thu, 22 Feb 2007 22:24:49 GMT CSeq: 49714 REGISTER User-Agent: CSCO/7 Contact: Content-Length: 0 Expires: 60 <-------------> [Feb 22 17:23:15] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:23:15] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK54a8e638 (58) [Feb 22 17:23:15] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: (40) [Feb 22 17:23:15] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (38) [Feb 22 17:23:15] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 000ff78d-ebb20002-237ee55f-6d451a9c@192.168.1.165 (58) [Feb 22 17:23:15] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Date: Thu, 22 Feb 2007 22:24:49 GMT (35) [Feb 22 17:23:15] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: CSeq: 49714 REGISTER (20) [Feb 22 17:23:15] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: CSCO/7 (18) [Feb 22 17:23:15] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Contact: (37) [Feb 22 17:23:15] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:23:15] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Expires: 60 (11) [Feb 22 17:23:15] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: (0) [Feb 22 17:23:15] --- (11 headers 0 lines) --- [Feb 22 17:23:15] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 000ff78d-ebb20002-237ee55f-6d451a9c@192.168.1.165 - REGISTER (No RTP) [Feb 22 17:23:15] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:23:15] Using latest REGISTER request as basis request [Feb 22 17:23:15] Sending to 192.168.1.165 : 5060 (no NAT) [Feb 22 17:23:15] <--- 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=z9hG4bK54a8e638;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20002-237ee55f-6d451a9c@192.168.1.165 CSeq: 49714 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:23:15] <--- 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=z9hG4bK54a8e638;received=192.168.1.165 From: To: ;tag=as3935e81a Call-ID: 000ff78d-ebb20002-237ee55f-6d451a9c@192.168.1.165 CSeq: 49714 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1559a9d9" Content-Length: 0 <------------> [Feb 22 17:23:15] Scheduling destruction of SIP dialog '000ff78d-ebb20002-237ee55f-6d451a9c@192.168.1.165' in 32000 ms (Method: REGISTER) [Feb 22 17:23:16] <--- 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=z9hG4bK70f54775 From: To: Call-ID: 000ff78d-ebb20002-237ee55f-6d451a9c@192.168.1.165 Date: Thu, 22 Feb 2007 22:24:49 GMT CSeq: 49715 REGISTER User-Agent: CSCO/7 Contact: Authorization: Digest username="510",realm="asterisk",uri="sip:24.123.23.170",response="79b48ebc4c446cbe47047be94d060b41",nonce="1559a9d9",algorithm=MD5 Content-Length: 0 Expires: 60 <-------------> [Feb 22 17:23:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:23:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK70f54775 (58) [Feb 22 17:23:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: (40) [Feb 22 17:23:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (38) [Feb 22 17:23:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 000ff78d-ebb20002-237ee55f-6d451a9c@192.168.1.165 (58) [Feb 22 17:23:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Date: Thu, 22 Feb 2007 22:24:49 GMT (35) [Feb 22 17:23:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: CSeq: 49715 REGISTER (20) [Feb 22 17:23:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: CSCO/7 (18) [Feb 22 17:23:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Contact: (37) [Feb 22 17:23:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Authorization: Digest username="510",realm="asterisk",uri="sip:24.123.23.170",response="79b48ebc4c446cbe47047be94d060b41",nonce="1559a9d9",algorithm=MD5 (152) [Feb 22 17:23:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Content-Length: 0 (17) [Feb 22 17:23:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Expires: 60 (11) [Feb 22 17:23:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: (0) [Feb 22 17:23:16] --- (12 headers 0 lines) --- [Feb 22 17:23:16] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:23:16] Using latest REGISTER request as basis request [Feb 22 17:23:16] Sending to 192.168.1.165 : 5060 (no NAT) [Feb 22 17:23:16] <--- 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=z9hG4bK70f54775;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20002-237ee55f-6d451a9c@192.168.1.165 CSeq: 49715 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:23:16] -- Saved useragent "CSCO/7" for peer 510 [Feb 22 17:23:16] <--- 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=z9hG4bK70f54775;received=192.168.1.165 From: To: ;tag=as3935e81a Call-ID: 000ff78d-ebb20002-237ee55f-6d451a9c@192.168.1.165 CSeq: 49715 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Thu, 22 Feb 2007 22:23:16 GMT Content-Length: 0 <------------> [Feb 22 17:23:16] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/510 [Feb 22 17:23:16] Scheduling destruction of SIP dialog '000ff78d-ebb20002-237ee55f-6d451a9c@192.168.1.165' in 32000 ms (Method: REGISTER) [Feb 22 17:23:16] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 510 [Feb 22 17:23:16] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 510 [Feb 22 17:23:16] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/510 - state 1 (Not in use) [Feb 22 17:23:16] DEBUG[14794]: app_queue.c:546 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. [Feb 22 17:23:16] <--- SIP read from 192.168.1.86:5062 ---> <-------------> [Feb 22 17:23:16] --- (0 headers 0 lines) Nat keepalive --- [Feb 22 17:23:16] <--- 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=z9hG4bK5fb92c60 From: To: Call-ID: 000ff78d-ebb20004-75d3ddb3-71096c5c@192.168.1.165 Date: Thu, 22 Feb 2007 22:24:50 GMT CSeq: 49690 REGISTER User-Agent: CSCO/7 Contact: Content-Length: 0 Expires: 60 <-------------> [Feb 22 17:23:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:23:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK5fb92c60 (58) [Feb 22 17:23:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: (40) [Feb 22 17:23:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (38) [Feb 22 17:23:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 000ff78d-ebb20004-75d3ddb3-71096c5c@192.168.1.165 (58) [Feb 22 17:23:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Date: Thu, 22 Feb 2007 22:24:50 GMT (35) [Feb 22 17:23:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: CSeq: 49690 REGISTER (20) [Feb 22 17:23:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: CSCO/7 (18) [Feb 22 17:23:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Contact: (37) [Feb 22 17:23:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:23:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Expires: 60 (11) [Feb 22 17:23:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: (0) [Feb 22 17:23:16] --- (11 headers 0 lines) --- [Feb 22 17:23:16] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 000ff78d-ebb20004-75d3ddb3-71096c5c@192.168.1.165 - REGISTER (No RTP) [Feb 22 17:23:16] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:23:16] Using latest REGISTER request as basis request [Feb 22 17:23:16] Sending to 192.168.1.165 : 5060 (no NAT) [Feb 22 17:23:16] <--- 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=z9hG4bK5fb92c60;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20004-75d3ddb3-71096c5c@192.168.1.165 CSeq: 49690 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:23:16] <--- 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=z9hG4bK5fb92c60;received=192.168.1.165 From: To: ;tag=as560345a9 Call-ID: 000ff78d-ebb20004-75d3ddb3-71096c5c@192.168.1.165 CSeq: 49690 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1ed6bf53" Content-Length: 0 <------------> [Feb 22 17:23:16] Scheduling destruction of SIP dialog '000ff78d-ebb20004-75d3ddb3-71096c5c@192.168.1.165' in 32000 ms (Method: REGISTER) [Feb 22 17:23:17] <--- 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=z9hG4bK4fb55546 From: To: Call-ID: 000ff78d-ebb20004-75d3ddb3-71096c5c@192.168.1.165 Date: Thu, 22 Feb 2007 22:24:50 GMT CSeq: 49691 REGISTER User-Agent: CSCO/7 Contact: Authorization: Digest username="512",realm="asterisk",uri="sip:24.123.23.170",response="2da49f0dfb7f09bd828463d346fd09b3",nonce="1ed6bf53",algorithm=MD5 Content-Length: 0 Expires: 60 <-------------> [Feb 22 17:23:17] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:23:17] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK4fb55546 (58) [Feb 22 17:23:17] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: (40) [Feb 22 17:23:17] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (38) [Feb 22 17:23:17] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 000ff78d-ebb20004-75d3ddb3-71096c5c@192.168.1.165 (58) [Feb 22 17:23:17] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Date: Thu, 22 Feb 2007 22:24:50 GMT (35) [Feb 22 17:23:17] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: CSeq: 49691 REGISTER (20) [Feb 22 17:23:17] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: CSCO/7 (18) [Feb 22 17:23:17] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Contact: (37) [Feb 22 17:23:17] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Authorization: Digest username="512",realm="asterisk",uri="sip:24.123.23.170",response="2da49f0dfb7f09bd828463d346fd09b3",nonce="1ed6bf53",algorithm=MD5 (152) [Feb 22 17:23:17] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Content-Length: 0 (17) [Feb 22 17:23:17] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Expires: 60 (11) [Feb 22 17:23:17] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: (0) [Feb 22 17:23:17] --- (12 headers 0 lines) --- [Feb 22 17:23:17] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:23:17] Using latest REGISTER request as basis request [Feb 22 17:23:17] Sending to 192.168.1.165 : 5060 (no NAT) [Feb 22 17:23:17] <--- 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=z9hG4bK4fb55546;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20004-75d3ddb3-71096c5c@192.168.1.165 CSeq: 49691 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:23:17] -- Saved useragent "CSCO/7" for peer 512 [Feb 22 17:23:17] <--- 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=z9hG4bK4fb55546;received=192.168.1.165 From: To: ;tag=as560345a9 Call-ID: 000ff78d-ebb20004-75d3ddb3-71096c5c@192.168.1.165 CSeq: 49691 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Thu, 22 Feb 2007 22:23:17 GMT Content-Length: 0 <------------> [Feb 22 17:23:17] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/512 [Feb 22 17:23:17] Scheduling destruction of SIP dialog '000ff78d-ebb20004-75d3ddb3-71096c5c@192.168.1.165' in 32000 ms (Method: REGISTER) [Feb 22 17:23:17] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 512 [Feb 22 17:23:17] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 512 [Feb 22 17:23:17] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/512 - state 1 (Not in use) [Feb 22 17:23:17] DEBUG[14795]: app_queue.c:546 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. [Feb 22 17:23:20] <--- 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=z9hG4bK3f6066e7 From: To: Call-ID: 000ff78d-ebb20006-6007ecd7-0ae287ec@192.168.1.165 Date: Thu, 22 Feb 2007 22:24:54 GMT CSeq: 49731 REGISTER User-Agent: CSCO/7 Contact: Content-Length: 0 Expires: 60 <-------------> [Feb 22 17:23:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:23:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK3f6066e7 (58) [Feb 22 17:23:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: (40) [Feb 22 17:23:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (38) [Feb 22 17:23:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 000ff78d-ebb20006-6007ecd7-0ae287ec@192.168.1.165 (58) [Feb 22 17:23:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Date: Thu, 22 Feb 2007 22:24:54 GMT (35) [Feb 22 17:23:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: CSeq: 49731 REGISTER (20) [Feb 22 17:23:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: CSCO/7 (18) [Feb 22 17:23:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Contact: (37) [Feb 22 17:23:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:23:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Expires: 60 (11) [Feb 22 17:23:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: (0) [Feb 22 17:23:20] --- (11 headers 0 lines) --- [Feb 22 17:23:20] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 000ff78d-ebb20006-6007ecd7-0ae287ec@192.168.1.165 - REGISTER (No RTP) [Feb 22 17:23:20] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:23:20] Using latest REGISTER request as basis request [Feb 22 17:23:20] Sending to 192.168.1.165 : 5060 (no NAT) [Feb 22 17:23:20] <--- 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=z9hG4bK3f6066e7;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20006-6007ecd7-0ae287ec@192.168.1.165 CSeq: 49731 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:23:20] <--- 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=z9hG4bK3f6066e7;received=192.168.1.165 From: To: ;tag=as4b80d833 Call-ID: 000ff78d-ebb20006-6007ecd7-0ae287ec@192.168.1.165 CSeq: 49731 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="48086aa3" Content-Length: 0 <------------> [Feb 22 17:23:20] Scheduling destruction of SIP dialog '000ff78d-ebb20006-6007ecd7-0ae287ec@192.168.1.165' in 32000 ms (Method: REGISTER) [Feb 22 17:23:20] <--- 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=z9hG4bK00544904 From: To: Call-ID: 000ff78d-ebb20006-6007ecd7-0ae287ec@192.168.1.165 Date: Thu, 22 Feb 2007 22:24:54 GMT CSeq: 49732 REGISTER User-Agent: CSCO/7 Contact: Authorization: Digest username="514",realm="asterisk",uri="sip:24.123.23.170",response="b26e82502570bd638a760ba0cf4ed1c2",nonce="48086aa3",algorithm=MD5 Content-Length: 0 Expires: 60 <-------------> [Feb 22 17:23:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:23:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK00544904 (58) [Feb 22 17:23:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: (40) [Feb 22 17:23:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (38) [Feb 22 17:23:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 000ff78d-ebb20006-6007ecd7-0ae287ec@192.168.1.165 (58) [Feb 22 17:23:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Date: Thu, 22 Feb 2007 22:24:54 GMT (35) [Feb 22 17:23:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: CSeq: 49732 REGISTER (20) [Feb 22 17:23:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: CSCO/7 (18) [Feb 22 17:23:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Contact: (37) [Feb 22 17:23:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Authorization: Digest username="514",realm="asterisk",uri="sip:24.123.23.170",response="b26e82502570bd638a760ba0cf4ed1c2",nonce="48086aa3",algorithm=MD5 (152) [Feb 22 17:23:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Content-Length: 0 (17) [Feb 22 17:23:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Expires: 60 (11) [Feb 22 17:23:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: (0) [Feb 22 17:23:20] --- (12 headers 0 lines) --- [Feb 22 17:23:20] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:23:20] Using latest REGISTER request as basis request [Feb 22 17:23:20] Sending to 192.168.1.165 : 5060 (no NAT) [Feb 22 17:23:20] <--- 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=z9hG4bK00544904;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20006-6007ecd7-0ae287ec@192.168.1.165 CSeq: 49732 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:23:20] <--- 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=z9hG4bK00544904;received=192.168.1.165 From: To: ;tag=as4b80d833 Call-ID: 000ff78d-ebb20006-6007ecd7-0ae287ec@192.168.1.165 CSeq: 49732 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Thu, 22 Feb 2007 22:23:20 GMT Content-Length: 0 <------------> [Feb 22 17:23:20] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/514 [Feb 22 17:23:20] Scheduling destruction of SIP dialog '000ff78d-ebb20006-6007ecd7-0ae287ec@192.168.1.165' in 32000 ms (Method: REGISTER) [Feb 22 17:23:20] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 514 [Feb 22 17:23:20] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 514 [Feb 22 17:23:20] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/514 - state 1 (Not in use) [Feb 22 17:23:20] DEBUG[14796]: app_queue.c:546 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. [Feb 22 17:23:21] <--- 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=z9hG4bKic6c3796181449f3593af0d10e10af5a0 Call-ID: 55ae664767e0@24.123.23.170 CSeq: 73552 REGISTER From: ;tag=714bb136cbf2b529b4f83cc500c3a840 To: Contact: ;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="7d7bf67e", algorithm=MD5, uri="sip:24.123.23.170", username="523", response="6338ea798de11652460bb029fdc78b35" <-------------> [Feb 22 17:23:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:23:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKic6c3796181449f3593af0d10e10af5a0 (82) [Feb 22 17:23:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: Call-ID: 55ae664767e0@24.123.23.170 (35) [Feb 22 17:23:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: CSeq: 73552 REGISTER (20) [Feb 22 17:23:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: ;tag=714bb136cbf2b529b4f83cc500c3a840 (66) [Feb 22 17:23:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: To: (27) [Feb 22 17:23:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Feb 22 17:23:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:23:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 22 17:23:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:23:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:23:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:23:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="7d7bf67e", algorithm=MD5, uri="sip:24.123.23.170", username="523", response="6338ea798de11652460bb029fdc78b35" (167) [Feb 22 17:23:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 13: (0) [Feb 22 17:23:21] --- (13 headers 0 lines) --- [Feb 22 17:23:21] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 55ae664767e0@24.123.23.170 - REGISTER (No RTP) [Feb 22 17:23:21] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:23:21] Using latest REGISTER request as basis request [Feb 22 17:23:21] Sending to 192.168.1.85 : 5060 (no NAT) [Feb 22 17:23:21] <--- 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=z9hG4bKic6c3796181449f3593af0d10e10af5a0;received=192.168.1.85 From: ;tag=714bb136cbf2b529b4f83cc500c3a840 To: Call-ID: 55ae664767e0@24.123.23.170 CSeq: 73552 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:23:21] <--- 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=z9hG4bKic6c3796181449f3593af0d10e10af5a0;received=192.168.1.85 From: ;tag=714bb136cbf2b529b4f83cc500c3a840 To: ;tag=as2b5f46b4 Call-ID: 55ae664767e0@24.123.23.170 CSeq: 73552 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="41b6d80e" Content-Length: 0 <------------> [Feb 22 17:23:21] Scheduling destruction of SIP dialog '55ae664767e0@24.123.23.170' in 32000 ms (Method: REGISTER) [Feb 22 17:23:21] <--- 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=z9hG4bKvf51a3bd0c0c23703d378226bac5c75d6 CSeq: 73553 REGISTER Call-ID: 55ae664767e0@24.123.23.170 From: ;tag=714bb136cbf2b529b4f83cc500c3a840 To: Contact: ;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="41b6d80e", algorithm=MD5, uri="sip:24.123.23.170", username="523", response="c62d8a32110b4fee2091d8efe1158013" <-------------> [Feb 22 17:23:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:23:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKvf51a3bd0c0c23703d378226bac5c75d6 (82) [Feb 22 17:23:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: CSeq: 73553 REGISTER (20) [Feb 22 17:23:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: Call-ID: 55ae664767e0@24.123.23.170 (35) [Feb 22 17:23:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: ;tag=714bb136cbf2b529b4f83cc500c3a840 (66) [Feb 22 17:23:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: To: (27) [Feb 22 17:23:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Feb 22 17:23:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:23:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 22 17:23:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:23:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:23:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:23:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="41b6d80e", algorithm=MD5, uri="sip:24.123.23.170", username="523", response="c62d8a32110b4fee2091d8efe1158013" (167) [Feb 22 17:23:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 13: (0) [Feb 22 17:23:21] --- (13 headers 0 lines) --- [Feb 22 17:23:21] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:23:21] Using latest REGISTER request as basis request [Feb 22 17:23:21] Sending to 192.168.1.85 : 5060 (no NAT) [Feb 22 17:23:21] <--- 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=z9hG4bKvf51a3bd0c0c23703d378226bac5c75d6;received=192.168.1.85 From: ;tag=714bb136cbf2b529b4f83cc500c3a840 To: Call-ID: 55ae664767e0@24.123.23.170 CSeq: 73553 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:23:21] <--- 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=z9hG4bKvf51a3bd0c0c23703d378226bac5c75d6;received=192.168.1.85 From: ;tag=714bb136cbf2b529b4f83cc500c3a840 To: ;tag=as2b5f46b4 Call-ID: 55ae664767e0@24.123.23.170 CSeq: 73553 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Thu, 22 Feb 2007 22:23:21 GMT Content-Length: 0 <------------> [Feb 22 17:23:21] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/523 [Feb 22 17:23:21] Scheduling destruction of SIP dialog '55ae664767e0@24.123.23.170' in 32000 ms (Method: REGISTER) [Feb 22 17:23:21] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 523 [Feb 22 17:23:21] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 523 [Feb 22 17:23:21] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/523 - state 1 (Not in use) [Feb 22 17:23:21] DEBUG[14797]: app_queue.c:546 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. [Feb 22 17:23:23] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Feb 22 17:23:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: OPTIONS sip:sip.broadvoice.com SIP/2.0 (38) [Feb 22 17:23:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK6a9f7dae;rport (64) [Feb 22 17:23:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: "asterisk" ;tag=as4758d2e2 (60) [Feb 22 17:23:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (28) [Feb 22 17:23:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Contact: (37) [Feb 22 17:23:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Call-ID: 02ccedfa5d22d5e46b7b5beb68f75269@24.123.23.170 (55) [Feb 22 17:23:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Feb 22 17:23:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Feb 22 17:23:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 22 17:23:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Date: Thu, 22 Feb 2007 22:23:23 GMT (35) [Feb 22 17:23:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Feb 22 17:23:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Supported: replaces (19) [Feb 22 17:23:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: Content-Length: 0 (17) [Feb 22 17:23:23] 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=z9hG4bK6a9f7dae;rport From: "asterisk" ;tag=as4758d2e2 To: Contact: Call-ID: 02ccedfa5d22d5e46b7b5beb68f75269@24.123.23.170 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 22 Feb 2007 22:23:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Feb 22 17:23:23] DEBUG[14767]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #137 [Feb 22 17:23:23] <--- SIP read from 147.135.12.128:5060 ---> SIP/2.0 200 OK Call-ID: 02ccedfa5d22d5e46b7b5beb68f75269@24.123.23.170 CSeq: 102 OPTIONS From: "asterisk" ;tag=as4758d2e2 To: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK6a9f7dae Supported: 100rel Allow: INVITE, BYE, ACK, OPTIONS, CANCEL, PRACK Accept: application/sdp Accept-Encoding: Accept-Language: en Content-Length: 0 <-------------> [Feb 22 17:23:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 22 17:23:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Call-ID: 02ccedfa5d22d5e46b7b5beb68f75269@24.123.23.170 (55) [Feb 22 17:23:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: CSeq: 102 OPTIONS (17) [Feb 22 17:23:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: From: "asterisk" ;tag=as4758d2e2 (60) [Feb 22 17:23:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: To: (28) [Feb 22 17:23:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK6a9f7dae (58) [Feb 22 17:23:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Supported: 100rel (17) [Feb 22 17:23:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: Allow: INVITE, BYE, ACK, OPTIONS, CANCEL, PRACK (47) [Feb 22 17:23:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Accept: application/sdp (23) [Feb 22 17:23:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Accept-Encoding: (17) [Feb 22 17:23:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Accept-Language: en (19) [Feb 22 17:23:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Content-Length: 0 (20) [Feb 22 17:23:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: (0) [Feb 22 17:23:23] --- (12 headers 0 lines) --- [Feb 22 17:23:23] DEBUG[14767]: chan_sip.c:2061 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #137 [Feb 22 17:23:23] DEBUG[14767]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '02ccedfa5d22d5e46b7b5beb68f75269@24.123.23.170' of Request 102: Match Not Found [Feb 22 17:23:23] Really destroying SIP dialog '02ccedfa5d22d5e46b7b5beb68f75269@24.123.23.170' Method: OPTIONS [Feb 22 17:23:24] DEBUG[14767]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '55ae6647666e@24.123.23.170' [Feb 22 17:23:24] DEBUG[14767]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 55ae6647666e@24.123.23.170 [Feb 22 17:23:24] Really destroying SIP dialog '55ae6647666e@24.123.23.170' Method: REGISTER [Feb 22 17:23:26] <--- 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=z9hG4bKh7b04af40edb81c73ff1f38e71f9fd9b5 Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 26670 REGISTER From: ;tag=21bad0a60809b2cfd7623801cdbc141e To: Contact: ;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="16bc22bb", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="7c983dc88bd821d841ca61fb16e7c488" <-------------> [Feb 22 17:23:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:23:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKh7b04af40edb81c73ff1f38e71f9fd9b5 (82) [Feb 22 17:23:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: Call-ID: 55ae66469f7f@24.123.23.170 (35) [Feb 22 17:23:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: CSeq: 26670 REGISTER (20) [Feb 22 17:23:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: ;tag=21bad0a60809b2cfd7623801cdbc141e (66) [Feb 22 17:23:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: To: (27) [Feb 22 17:23:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Feb 22 17:23:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:23:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 22 17:23:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:23:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:23:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:23:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="16bc22bb", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="7c983dc88bd821d841ca61fb16e7c488" (167) [Feb 22 17:23:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 13: (0) [Feb 22 17:23:26] --- (13 headers 0 lines) --- [Feb 22 17:23:26] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 55ae66469f7f@24.123.23.170 - REGISTER (No RTP) [Feb 22 17:23:26] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:23:26] Using latest REGISTER request as basis request [Feb 22 17:23:26] Sending to 192.168.1.93 : 5060 (no NAT) [Feb 22 17:23:26] <--- 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=z9hG4bKh7b04af40edb81c73ff1f38e71f9fd9b5;received=192.168.1.93 From: ;tag=21bad0a60809b2cfd7623801cdbc141e To: Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 26670 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:23:26] <--- 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=z9hG4bKh7b04af40edb81c73ff1f38e71f9fd9b5;received=192.168.1.93 From: ;tag=21bad0a60809b2cfd7623801cdbc141e To: ;tag=as1ecaa06e Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 26670 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4ee04135" Content-Length: 0 <------------> [Feb 22 17:23:26] Scheduling destruction of SIP dialog '55ae66469f7f@24.123.23.170' in 32000 ms (Method: REGISTER) [Feb 22 17:23:26] <--- 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=z9hG4bKu8655f67dc0344c3c31e608f14017e8a7 CSeq: 26671 REGISTER Call-ID: 55ae66469f7f@24.123.23.170 From: ;tag=21bad0a60809b2cfd7623801cdbc141e To: Contact: ;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="4ee04135", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="6df9ba88fbf3ecee38524e2f12d5dace" <-------------> [Feb 22 17:23:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:23:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKu8655f67dc0344c3c31e608f14017e8a7 (82) [Feb 22 17:23:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: CSeq: 26671 REGISTER (20) [Feb 22 17:23:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: Call-ID: 55ae66469f7f@24.123.23.170 (35) [Feb 22 17:23:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: ;tag=21bad0a60809b2cfd7623801cdbc141e (66) [Feb 22 17:23:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: To: (27) [Feb 22 17:23:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Feb 22 17:23:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:23:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 22 17:23:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:23:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:23:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:23:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="4ee04135", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="6df9ba88fbf3ecee38524e2f12d5dace" (167) [Feb 22 17:23:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 13: (0) [Feb 22 17:23:26] --- (13 headers 0 lines) --- [Feb 22 17:23:26] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:23:26] Using latest REGISTER request as basis request [Feb 22 17:23:26] Sending to 192.168.1.93 : 5060 (no NAT) [Feb 22 17:23:26] <--- 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=z9hG4bKu8655f67dc0344c3c31e608f14017e8a7;received=192.168.1.93 From: ;tag=21bad0a60809b2cfd7623801cdbc141e To: Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 26671 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:23:26] <--- 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=z9hG4bKu8655f67dc0344c3c31e608f14017e8a7;received=192.168.1.93 From: ;tag=21bad0a60809b2cfd7623801cdbc141e To: ;tag=as1ecaa06e Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 26671 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Thu, 22 Feb 2007 22:23:26 GMT Content-Length: 0 <------------> [Feb 22 17:23:26] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/528 [Feb 22 17:23:26] Scheduling destruction of SIP dialog '55ae66469f7f@24.123.23.170' in 32000 ms (Method: REGISTER) [Feb 22 17:23:26] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 528 [Feb 22 17:23:26] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 528 [Feb 22 17:23:26] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/528 - state 1 (Not in use) [Feb 22 17:23:26] DEBUG[14798]: app_queue.c:546 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. [Feb 22 17:23:26] DEBUG[14767]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '000ff78d-ebb20007-74946e6d-3266b768@192.168.1.165' [Feb 22 17:23:26] DEBUG[14767]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 000ff78d-ebb20007-74946e6d-3266b768@192.168.1.165 [Feb 22 17:23:26] Really destroying SIP dialog '000ff78d-ebb20007-74946e6d-3266b768@192.168.1.165' Method: REGISTER [Feb 22 17:23:26] <--- SIP read from 192.168.1.86:5060 ---> <-------------> [Feb 22 17:23:26] --- (0 headers 0 lines) Nat keepalive --- [Feb 22 17:23:27] DEBUG[14767]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '55ae664696c0@24.123.23.170' [Feb 22 17:23:27] DEBUG[14767]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 55ae664696c0@24.123.23.170 [Feb 22 17:23:27] Really destroying SIP dialog '55ae664696c0@24.123.23.170' Method: REGISTER [Feb 22 17:23:28] <--- 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=z9hG4bKod4a16257e716c68759861aa24c2f8bb2 Call-ID: 55ae66476793@24.123.23.170 CSeq: 108462 REGISTER From: ;tag=1c3b4e06d31058af35aae83bf8827d18 To: Contact: ;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="79ee4023", algorithm=MD5, uri="sip:24.123.23.170", username="525", response="8d82c1f102495c41d356d8637be992a8" <-------------> [Feb 22 17:23:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:23:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bKod4a16257e716c68759861aa24c2f8bb2 (82) [Feb 22 17:23:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: Call-ID: 55ae66476793@24.123.23.170 (35) [Feb 22 17:23:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: CSeq: 108462 REGISTER (21) [Feb 22 17:23:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: ;tag=1c3b4e06d31058af35aae83bf8827d18 (66) [Feb 22 17:23:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: To: (27) [Feb 22 17:23:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Feb 22 17:23:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:23:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 22 17:23:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:23:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:23:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:23:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="79ee4023", algorithm=MD5, uri="sip:24.123.23.170", username="525", response="8d82c1f102495c41d356d8637be992a8" (167) [Feb 22 17:23:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 13: (0) [Feb 22 17:23:28] --- (13 headers 0 lines) --- [Feb 22 17:23:28] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 55ae66476793@24.123.23.170 - REGISTER (No RTP) [Feb 22 17:23:28] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:23:28] Using latest REGISTER request as basis request [Feb 22 17:23:28] Sending to 192.168.1.76 : 5060 (no NAT) [Feb 22 17:23:28] <--- 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=z9hG4bKod4a16257e716c68759861aa24c2f8bb2;received=192.168.1.76 From: ;tag=1c3b4e06d31058af35aae83bf8827d18 To: Call-ID: 55ae66476793@24.123.23.170 CSeq: 108462 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:23:28] <--- 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=z9hG4bKod4a16257e716c68759861aa24c2f8bb2;received=192.168.1.76 From: ;tag=1c3b4e06d31058af35aae83bf8827d18 To: ;tag=as7cde6d6d Call-ID: 55ae66476793@24.123.23.170 CSeq: 108462 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="101b6db8" Content-Length: 0 <------------> [Feb 22 17:23:28] Scheduling destruction of SIP dialog '55ae66476793@24.123.23.170' in 32000 ms (Method: REGISTER) [Feb 22 17:23:28] <--- SIP read from 68.58.36.157:5060 ---> <-------------> [Feb 22 17:23:28] --- (0 headers 0 lines) Nat keepalive --- [Feb 22 17:23:28] <--- 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=z9hG4bKp473e3e47730318494a789d3c22e88716 CSeq: 108463 REGISTER Call-ID: 55ae66476793@24.123.23.170 From: ;tag=1c3b4e06d31058af35aae83bf8827d18 To: Contact: ;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="101b6db8", algorithm=MD5, uri="sip:24.123.23.170", username="525", response="fbdcf2731f8b245232f034935deddd98" <-------------> [Feb 22 17:23:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:23:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bKp473e3e47730318494a789d3c22e88716 (82) [Feb 22 17:23:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: CSeq: 108463 REGISTER (21) [Feb 22 17:23:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: Call-ID: 55ae66476793@24.123.23.170 (35) [Feb 22 17:23:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: ;tag=1c3b4e06d31058af35aae83bf8827d18 (66) [Feb 22 17:23:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: To: (27) [Feb 22 17:23:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Feb 22 17:23:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:23:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 22 17:23:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:23:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:23:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:23:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="101b6db8", algorithm=MD5, uri="sip:24.123.23.170", username="525", response="fbdcf2731f8b245232f034935deddd98" (167) [Feb 22 17:23:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 13: (0) [Feb 22 17:23:28] --- (13 headers 0 lines) --- [Feb 22 17:23:28] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:23:28] Using latest REGISTER request as basis request [Feb 22 17:23:28] Sending to 192.168.1.76 : 5060 (no NAT) [Feb 22 17:23:28] <--- 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=z9hG4bKp473e3e47730318494a789d3c22e88716;received=192.168.1.76 From: ;tag=1c3b4e06d31058af35aae83bf8827d18 To: Call-ID: 55ae66476793@24.123.23.170 CSeq: 108463 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:23:28] <--- 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=z9hG4bKp473e3e47730318494a789d3c22e88716;received=192.168.1.76 From: ;tag=1c3b4e06d31058af35aae83bf8827d18 To: ;tag=as7cde6d6d Call-ID: 55ae66476793@24.123.23.170 CSeq: 108463 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Thu, 22 Feb 2007 22:23:28 GMT Content-Length: 0 <------------> [Feb 22 17:23:28] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/525 [Feb 22 17:23:28] Scheduling destruction of SIP dialog '55ae66476793@24.123.23.170' in 32000 ms (Method: REGISTER) [Feb 22 17:23:28] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 525 [Feb 22 17:23:28] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 525 [Feb 22 17:23:28] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/525 - state 1 (Not in use) [Feb 22 17:23:28] DEBUG[14802]: app_queue.c:546 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. [Feb 22 17:23:30] DEBUG[14767]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '000ff78d-ebb20003-42dc795d-3e449dd6@192.168.1.165' [Feb 22 17:23:30] DEBUG[14767]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 000ff78d-ebb20003-42dc795d-3e449dd6@192.168.1.165 [Feb 22 17:23:30] Really destroying SIP dialog '000ff78d-ebb20003-42dc795d-3e449dd6@192.168.1.165' Method: REGISTER [Feb 22 17:23:31] NOTICE[14767]: chan_sip.c:7055 sip_reregister: -- Re-registration for 3173241052@sip.broadvoice.com [Feb 22 17:23:31] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 42fcb9a2205a226e1657915a20f1c05a@24.123.23.170 - REGISTER (No RTP) [Feb 22 17:23:31] DEBUG[14767]: chan_sip.c:7214 transmit_register: Scheduled a registration timeout for sip.broadvoice.com id #146 [Feb 22 17:23:31] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:sip.broadvoice.com SIP/2.0 (39) [Feb 22 17:23:31] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK287206e4;rport (64) [Feb 22 17:23:31] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: ;tag=as1dd19b38 (56) [Feb 22 17:23:31] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (39) [Feb 22 17:23:31] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 42fcb9a2205a226e1657915a20f1c05a@24.123.23.170 (55) [Feb 22 17:23:31] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: CSeq: 105 REGISTER (18) [Feb 22 17:23:31] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Feb 22 17:23:31] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: Max-Forwards: 70 (16) [Feb 22 17:23:31] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Expires: 120 (12) [Feb 22 17:23:31] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Contact: (39) [Feb 22 17:23:31] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Event: registration (19) [Feb 22 17:23:31] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Content-Length: 0 (17) [Feb 22 17:23:31] REGISTER 12 headers, 0 lines [Feb 22 17:23:31] REGISTER attempt 1 to 3173241052@sip.broadvoice.com [Feb 22 17:23:31] 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=z9hG4bK287206e4;rport From: ;tag=as1dd19b38 To: Call-ID: 42fcb9a2205a226e1657915a20f1c05a@24.123.23.170 CSeq: 105 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: Event: registration Content-Length: 0 --- [Feb 22 17:23:31] DEBUG[14767]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #147 [Feb 22 17:23:31] <--- SIP read from 147.135.12.128:5060 ---> SIP/2.0 200 OK Call-ID: 42fcb9a2205a226e1657915a20f1c05a@24.123.23.170 CSeq: 105 REGISTER From: ;tag=as1dd19b38 To: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK287206e4 Contact: Expires: 30 Event: registration Content-Length: 0 <-------------> [Feb 22 17:23:31] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 22 17:23:31] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Call-ID: 42fcb9a2205a226e1657915a20f1c05a@24.123.23.170 (55) [Feb 22 17:23:31] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: CSeq: 105 REGISTER (18) [Feb 22 17:23:31] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: From: ;tag=as1dd19b38 (56) [Feb 22 17:23:31] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: To: (39) [Feb 22 17:23:31] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK287206e4 (58) [Feb 22 17:23:31] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: (39) [Feb 22 17:23:31] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: Expires: 30 (11) [Feb 22 17:23:31] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Event: registration (19) [Feb 22 17:23:31] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (20) [Feb 22 17:23:31] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: (0) [Feb 22 17:23:31] --- (10 headers 0 lines) --- [Feb 22 17:23:31] DEBUG[14767]: chan_sip.c:2061 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #147 [Feb 22 17:23:31] DEBUG[14767]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '42fcb9a2205a226e1657915a20f1c05a@24.123.23.170' of Request 105: Match Not Found [Feb 22 17:23:31] DEBUG[14767]: chan_sip.c:11906 handle_response_register: Registration successful [Feb 22 17:23:31] DEBUG[14767]: chan_sip.c:11909 handle_response_register: Cancelling timeout 146 [Feb 22 17:23:31] Scheduling destruction of SIP dialog '42fcb9a2205a226e1657915a20f1c05a@24.123.23.170' in 32000 ms (Method: REGISTER) [Feb 22 17:23:31] NOTICE[14767]: chan_sip.c:11961 handle_response_register: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 23 s) [Feb 22 17:23:32] DEBUG[14763]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel IAX2/651 [Feb 22 17:23:32] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for IAX2 - 651 [Feb 22 17:23:32] DEBUG[14741]: chan_iax2.c:9600 iax2_devicestate: Checking device state for device 651 [Feb 22 17:23:32] DEBUG[14741]: chan_iax2.c:9608 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 651? addr=-1459509056, defaddr=0 maxms=0, lastms=0 [Feb 22 17:23:32] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for IAX2/651 - state 1 (Not in use) [Feb 22 17:23:32] DEBUG[14803]: app_queue.c:546 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. [Feb 22 17:23:35] DEBUG[14767]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '000ff78d-ebb20005-4fe621ee-067da175@192.168.1.165' [Feb 22 17:23:35] DEBUG[14767]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 000ff78d-ebb20005-4fe621ee-067da175@192.168.1.165 [Feb 22 17:23:35] Really destroying SIP dialog '000ff78d-ebb20005-4fe621ee-067da175@192.168.1.165' Method: REGISTER [Feb 22 17:23:35] DEBUG[14754]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel IAX2/614 [Feb 22 17:23:35] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for IAX2 - 614 [Feb 22 17:23:35] DEBUG[14741]: chan_iax2.c:9600 iax2_devicestate: Checking device state for device 614 [Feb 22 17:23:35] DEBUG[14741]: chan_iax2.c:9608 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 614? addr=-1493063488, defaddr=0 maxms=0, lastms=0 [Feb 22 17:23:35] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for IAX2/614 - state 1 (Not in use) [Feb 22 17:23:35] DEBUG[14805]: app_queue.c:546 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. [Feb 22 17:23:36] <--- SIP read from 192.168.1.86:5062 ---> <-------------> [Feb 22 17:23:36] --- (0 headers 0 lines) Nat keepalive --- [Feb 22 17:23:39] DEBUG[14767]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '42fcb9a2205a226e1657915a20f1c05a@24.123.23.170' [Feb 22 17:23:39] DEBUG[14767]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 42fcb9a2205a226e1657915a20f1c05a@24.123.23.170 [Feb 22 17:23:39] Really destroying SIP dialog '42fcb9a2205a226e1657915a20f1c05a@24.123.23.170' Method: REGISTER [Feb 22 17:23:40] <--- 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=z9hG4bKv83cd9b5949c179873086b13d3ba53242 Call-ID: 55ae66469f11@24.123.23.170 CSeq: 33544 REGISTER From: ;tag=8c81e28f14d76f912dff6f28eb488f5b To: Contact: ;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="090c0bbb", algorithm=MD5, uri="sip:24.123.23.170", username="522", response="1613d16543f28039f9e4de2a44d892e5" <-------------> [Feb 22 17:23:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:23:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.95:5060;branch=z9hG4bKv83cd9b5949c179873086b13d3ba53242 (82) [Feb 22 17:23:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: Call-ID: 55ae66469f11@24.123.23.170 (35) [Feb 22 17:23:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: CSeq: 33544 REGISTER (20) [Feb 22 17:23:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: ;tag=8c81e28f14d76f912dff6f28eb488f5b (66) [Feb 22 17:23:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: To: (27) [Feb 22 17:23:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Feb 22 17:23:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:23:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 22 17:23:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:23:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:23:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:23:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="090c0bbb", algorithm=MD5, uri="sip:24.123.23.170", username="522", response="1613d16543f28039f9e4de2a44d892e5" (167) [Feb 22 17:23:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 13: (0) [Feb 22 17:23:40] --- (13 headers 0 lines) --- [Feb 22 17:23:40] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 55ae66469f11@24.123.23.170 - REGISTER (No RTP) [Feb 22 17:23:40] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:23:40] Using latest REGISTER request as basis request [Feb 22 17:23:40] Sending to 192.168.1.95 : 5060 (no NAT) [Feb 22 17:23:40] <--- 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=z9hG4bKv83cd9b5949c179873086b13d3ba53242;received=192.168.1.95 From: ;tag=8c81e28f14d76f912dff6f28eb488f5b To: Call-ID: 55ae66469f11@24.123.23.170 CSeq: 33544 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:23:40] <--- 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=z9hG4bKv83cd9b5949c179873086b13d3ba53242;received=192.168.1.95 From: ;tag=8c81e28f14d76f912dff6f28eb488f5b To: ;tag=as0d51c27e Call-ID: 55ae66469f11@24.123.23.170 CSeq: 33544 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="57cac9ba" Content-Length: 0 <------------> [Feb 22 17:23:40] Scheduling destruction of SIP dialog '55ae66469f11@24.123.23.170' in 32000 ms (Method: REGISTER) [Feb 22 17:23:40] <--- 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=z9hG4bKd0fcc704e91bfe127d0092fc6303f70ac CSeq: 33545 REGISTER Call-ID: 55ae66469f11@24.123.23.170 From: ;tag=8c81e28f14d76f912dff6f28eb488f5b To: Contact: ;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="57cac9ba", algorithm=MD5, uri="sip:24.123.23.170", username="522", response="3e037aefd24e5ff3815a099fac8aa29b" <-------------> [Feb 22 17:23:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:23:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.95:5060;branch=z9hG4bKd0fcc704e91bfe127d0092fc6303f70ac (82) [Feb 22 17:23:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: CSeq: 33545 REGISTER (20) [Feb 22 17:23:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: Call-ID: 55ae66469f11@24.123.23.170 (35) [Feb 22 17:23:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: ;tag=8c81e28f14d76f912dff6f28eb488f5b (66) [Feb 22 17:23:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: To: (27) [Feb 22 17:23:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Feb 22 17:23:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:23:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 22 17:23:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:23:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:23:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:23:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="57cac9ba", algorithm=MD5, uri="sip:24.123.23.170", username="522", response="3e037aefd24e5ff3815a099fac8aa29b" (167) [Feb 22 17:23:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 13: (0) [Feb 22 17:23:40] --- (13 headers 0 lines) --- [Feb 22 17:23:40] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:23:40] Using latest REGISTER request as basis request [Feb 22 17:23:40] Sending to 192.168.1.95 : 5060 (no NAT) [Feb 22 17:23:40] <--- 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=z9hG4bKd0fcc704e91bfe127d0092fc6303f70ac;received=192.168.1.95 From: ;tag=8c81e28f14d76f912dff6f28eb488f5b To: Call-ID: 55ae66469f11@24.123.23.170 CSeq: 33545 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:23:40] <--- 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=z9hG4bKd0fcc704e91bfe127d0092fc6303f70ac;received=192.168.1.95 From: ;tag=8c81e28f14d76f912dff6f28eb488f5b To: ;tag=as0d51c27e Call-ID: 55ae66469f11@24.123.23.170 CSeq: 33545 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Thu, 22 Feb 2007 22:23:40 GMT Content-Length: 0 <------------> [Feb 22 17:23:40] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/522 [Feb 22 17:23:40] Scheduling destruction of SIP dialog '55ae66469f11@24.123.23.170' in 32000 ms (Method: REGISTER) [Feb 22 17:23:40] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 522 [Feb 22 17:23:40] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 522 [Feb 22 17:23:40] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/522 - state 1 (Not in use) [Feb 22 17:23:40] DEBUG[14806]: app_queue.c:546 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. [Feb 22 17:23:46] <--- SIP read from 192.168.1.86:5060 ---> <-------------> [Feb 22 17:23:46] --- (0 headers 0 lines) Nat keepalive --- [Feb 22 17:23:46] DEBUG[14758]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel IAX2/606 [Feb 22 17:23:46] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for IAX2 - 606 [Feb 22 17:23:46] DEBUG[14741]: chan_iax2.c:9600 iax2_devicestate: Checking device state for device 606 [Feb 22 17:23:46] DEBUG[14741]: chan_iax2.c:9608 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 606? addr=-1509840704, defaddr=0 maxms=0, lastms=0 [Feb 22 17:23:46] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for IAX2/606 - state 1 (Not in use) [Feb 22 17:23:46] DEBUG[14808]: app_queue.c:546 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. [Feb 22 17:23:48] DEBUG[14767]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '000ff78d-ebb20002-237ee55f-6d451a9c@192.168.1.165' [Feb 22 17:23:48] DEBUG[14767]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 000ff78d-ebb20002-237ee55f-6d451a9c@192.168.1.165 [Feb 22 17:23:48] Really destroying SIP dialog '000ff78d-ebb20002-237ee55f-6d451a9c@192.168.1.165' Method: REGISTER [Feb 22 17:23:48] <--- SIP read from 68.58.36.157:5060 ---> <-------------> [Feb 22 17:23:48] --- (0 headers 0 lines) Nat keepalive --- [Feb 22 17:23:49] DEBUG[14767]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '000ff78d-ebb20004-75d3ddb3-71096c5c@192.168.1.165' [Feb 22 17:23:49] DEBUG[14767]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 000ff78d-ebb20004-75d3ddb3-71096c5c@192.168.1.165 [Feb 22 17:23:49] Really destroying SIP dialog '000ff78d-ebb20004-75d3ddb3-71096c5c@192.168.1.165' Method: REGISTER [Feb 22 17:23:50] DEBUG[14761]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel IAX2/616 [Feb 22 17:23:50] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for IAX2 - 616 [Feb 22 17:23:50] DEBUG[14741]: chan_iax2.c:9600 iax2_devicestate: Checking device state for device 616 [Feb 22 17:23:50] DEBUG[14741]: chan_iax2.c:9608 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 616? addr=-1891793590, defaddr=0 maxms=0, lastms=0 [Feb 22 17:23:50] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for IAX2/616 - state 1 (Not in use) [Feb 22 17:23:50] DEBUG[14809]: app_queue.c:546 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. [Feb 22 17:23:52] <--- 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=z9hG4bKt27d1dee27f495f81e45b88494b3e2c2c Call-ID: 55ae6647666e@24.123.23.170 CSeq: 64474 REGISTER From: ;tag=9262cb7040e3377f966eb467c1ba09c7 To: Contact: ;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="39b1f81b", algorithm=MD5, uri="sip:24.123.23.170", username="526", response="5469d046dc161dfca54960cb9bad49ed" <-------------> [Feb 22 17:23:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:23:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bKt27d1dee27f495f81e45b88494b3e2c2c (82) [Feb 22 17:23:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: Call-ID: 55ae6647666e@24.123.23.170 (35) [Feb 22 17:23:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: CSeq: 64474 REGISTER (20) [Feb 22 17:23:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: ;tag=9262cb7040e3377f966eb467c1ba09c7 (66) [Feb 22 17:23:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: To: (27) [Feb 22 17:23:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Feb 22 17:23:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:23:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 22 17:23:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:23:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:23:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:23:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="39b1f81b", algorithm=MD5, uri="sip:24.123.23.170", username="526", response="5469d046dc161dfca54960cb9bad49ed" (167) [Feb 22 17:23:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 13: (0) [Feb 22 17:23:52] --- (13 headers 0 lines) --- [Feb 22 17:23:52] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 55ae6647666e@24.123.23.170 - REGISTER (No RTP) [Feb 22 17:23:52] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:23:52] Using latest REGISTER request as basis request [Feb 22 17:23:52] Sending to 192.168.1.66 : 5060 (no NAT) [Feb 22 17:23:52] <--- 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=z9hG4bKt27d1dee27f495f81e45b88494b3e2c2c;received=192.168.1.66 From: ;tag=9262cb7040e3377f966eb467c1ba09c7 To: Call-ID: 55ae6647666e@24.123.23.170 CSeq: 64474 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:23:52] <--- 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=z9hG4bKt27d1dee27f495f81e45b88494b3e2c2c;received=192.168.1.66 From: ;tag=9262cb7040e3377f966eb467c1ba09c7 To: ;tag=as126dec6c Call-ID: 55ae6647666e@24.123.23.170 CSeq: 64474 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="50a4fb98" Content-Length: 0 <------------> [Feb 22 17:23:52] Scheduling destruction of SIP dialog '55ae6647666e@24.123.23.170' in 32000 ms (Method: REGISTER) [Feb 22 17:23:52] DEBUG[14767]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '000ff78d-ebb20006-6007ecd7-0ae287ec@192.168.1.165' [Feb 22 17:23:52] DEBUG[14767]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 000ff78d-ebb20006-6007ecd7-0ae287ec@192.168.1.165 [Feb 22 17:23:52] Really destroying SIP dialog '000ff78d-ebb20006-6007ecd7-0ae287ec@192.168.1.165' Method: REGISTER [Feb 22 17:23:52] <--- 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=z9hG4bKy9222b47cb33fa1f6517880ae52fad041 CSeq: 64475 REGISTER Call-ID: 55ae6647666e@24.123.23.170 From: ;tag=9262cb7040e3377f966eb467c1ba09c7 To: Contact: ;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="50a4fb98", algorithm=MD5, uri="sip:24.123.23.170", username="526", response="f47b7922eb51965b0a9127c7f5f7ae7a" <-------------> [Feb 22 17:23:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:23:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bKy9222b47cb33fa1f6517880ae52fad041 (82) [Feb 22 17:23:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: CSeq: 64475 REGISTER (20) [Feb 22 17:23:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: Call-ID: 55ae6647666e@24.123.23.170 (35) [Feb 22 17:23:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: ;tag=9262cb7040e3377f966eb467c1ba09c7 (66) [Feb 22 17:23:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: To: (27) [Feb 22 17:23:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Feb 22 17:23:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:23:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 22 17:23:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:23:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:23:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:23:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="50a4fb98", algorithm=MD5, uri="sip:24.123.23.170", username="526", response="f47b7922eb51965b0a9127c7f5f7ae7a" (167) [Feb 22 17:23:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 13: (0) [Feb 22 17:23:52] --- (13 headers 0 lines) --- [Feb 22 17:23:52] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:23:52] Using latest REGISTER request as basis request [Feb 22 17:23:52] Sending to 192.168.1.66 : 5060 (no NAT) [Feb 22 17:23:52] <--- 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=z9hG4bKy9222b47cb33fa1f6517880ae52fad041;received=192.168.1.66 From: ;tag=9262cb7040e3377f966eb467c1ba09c7 To: Call-ID: 55ae6647666e@24.123.23.170 CSeq: 64475 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:23:52] <--- 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=z9hG4bKy9222b47cb33fa1f6517880ae52fad041;received=192.168.1.66 From: ;tag=9262cb7040e3377f966eb467c1ba09c7 To: ;tag=as126dec6c Call-ID: 55ae6647666e@24.123.23.170 CSeq: 64475 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Thu, 22 Feb 2007 22:23:52 GMT Content-Length: 0 <------------> [Feb 22 17:23:52] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/526 [Feb 22 17:23:52] Scheduling destruction of SIP dialog '55ae6647666e@24.123.23.170' in 32000 ms (Method: REGISTER) [Feb 22 17:23:52] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 526 [Feb 22 17:23:52] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 526 [Feb 22 17:23:52] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/526 - state 1 (Not in use) [Feb 22 17:23:52] DEBUG[14810]: app_queue.c:546 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. [Feb 22 17:23:53] <--- SIP read from 192.168.1.91:5060 ---> REGISTER sip:24.123.23.170:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.91:5060;rport;branch=z9hG4bK1672211442 From: ;tag=246616177 To: Call-ID: 2731658924@192.168.1.91 CSeq: 1 REGISTER Contact: ;action=proxy max-forwards: 70 Expires: 60 user-agent: UTSTARCOM F3000/Device ID-F3000_TEST Content-Length: 0 <-------------> [Feb 22 17:23:53] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170:5060 SIP/2.0 (39) [Feb 22 17:23:53] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.91:5060;rport;branch=z9hG4bK1672211442 (65) [Feb 22 17:23:53] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: ;tag=246616177 (48) [Feb 22 17:23:53] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (32) [Feb 22 17:23:53] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 2731658924@192.168.1.91 (32) [Feb 22 17:23:53] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: CSeq: 1 REGISTER (16) [Feb 22 17:23:53] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: ;action=proxy (49) [Feb 22 17:23:53] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: max-forwards: 70 (16) [Feb 22 17:23:53] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Expires: 60 (11) [Feb 22 17:23:53] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: user-agent: UTSTARCOM F3000/Device ID-F3000_TEST (48) [Feb 22 17:23:53] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Content-Length: 0 (17) [Feb 22 17:23:53] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: (0) [Feb 22 17:23:53] --- (11 headers 0 lines) --- [Feb 22 17:23:53] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 2731658924@192.168.1.91 - REGISTER (No RTP) [Feb 22 17:23:53] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:23:53] Using latest REGISTER request as basis request [Feb 22 17:23:53] Sending to 192.168.1.91 : 5060 (NAT) [Feb 22 17:23:53] <--- Transmitting (NAT) to 192.168.1.91:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.91:5060;branch=z9hG4bK1672211442;received=192.168.1.91;rport=5060 From: ;tag=246616177 To: Call-ID: 2731658924@192.168.1.91 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:23:53] <--- Transmitting (NAT) to 192.168.1.91:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.91:5060;branch=z9hG4bK1672211442;received=192.168.1.91;rport=5060 From: ;tag=246616177 To: ;tag=as3a0805aa Call-ID: 2731658924@192.168.1.91 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0cada832" Content-Length: 0 <------------> [Feb 22 17:23:53] Scheduling destruction of SIP dialog '2731658924@192.168.1.91' in 32000 ms (Method: REGISTER) [Feb 22 17:23:53] DEBUG[14767]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '55ae664767e0@24.123.23.170' [Feb 22 17:23:53] DEBUG[14767]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 55ae664767e0@24.123.23.170 [Feb 22 17:23:53] Really destroying SIP dialog '55ae664767e0@24.123.23.170' Method: REGISTER [Feb 22 17:23:53] <--- SIP read from 192.168.1.91:5060 ---> REGISTER sip:24.123.23.170:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.91:5060;rport;branch=z9hG4bK889693480 From: ;tag=3120892927 To: Call-ID: 2731658924@192.168.1.91 CSeq: 2 REGISTER Contact: ;action=proxy Authorization: Digest username="550", realm="asterisk", nonce="0cada832", uri="sip:24.123.23.170:5060", response="e4ad612eb4825f88c4cca40687bea858", algorithm=MD5 max-forwards: 70 Expires: 60 user-agent: UTSTARCOM F3000/Device ID-F3000_TEST Content-Length: 0 <-------------> [Feb 22 17:23:53] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170:5060 SIP/2.0 (39) [Feb 22 17:23:53] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.91:5060;rport;branch=z9hG4bK889693480 (64) [Feb 22 17:23:53] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: ;tag=3120892927 (49) [Feb 22 17:23:53] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (32) [Feb 22 17:23:53] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 2731658924@192.168.1.91 (32) [Feb 22 17:23:53] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: CSeq: 2 REGISTER (16) [Feb 22 17:23:53] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: ;action=proxy (49) [Feb 22 17:23:53] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: Authorization: Digest username="550", realm="asterisk", nonce="0cada832", uri="sip:24.123.23.170:5060", response="e4ad612eb4825f88c4cca40687bea858", algorithm=MD5 (162) [Feb 22 17:23:53] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: max-forwards: 70 (16) [Feb 22 17:23:53] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Expires: 60 (11) [Feb 22 17:23:53] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: user-agent: UTSTARCOM F3000/Device ID-F3000_TEST (48) [Feb 22 17:23:53] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Content-Length: 0 (17) [Feb 22 17:23:53] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: (0) [Feb 22 17:23:53] --- (12 headers 0 lines) --- [Feb 22 17:23:53] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:23:53] Using latest REGISTER request as basis request [Feb 22 17:23:53] Sending to 192.168.1.91 : 5060 (NAT) [Feb 22 17:23:53] <--- Transmitting (NAT) to 192.168.1.91:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.91:5060;branch=z9hG4bK889693480;received=192.168.1.91;rport=5060 From: ;tag=3120892927 To: Call-ID: 2731658924@192.168.1.91 CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:23:53] -- Registered SIP '550' at 192.168.1.91 port 5060 expires 60 [Feb 22 17:23:53] -- Saved useragent "UTSTARCOM F3000/Device ID-F3000_TEST" for peer 550 [Feb 22 17:23:53] <--- Transmitting (NAT) to 192.168.1.91:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.91:5060;branch=z9hG4bK889693480;received=192.168.1.91;rport=5060 From: ;tag=3120892927 To: ;tag=as3a0805aa Call-ID: 2731658924@192.168.1.91 CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Thu, 22 Feb 2007 22:23:53 GMT Content-Length: 0 <------------> [Feb 22 17:23:53] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/550 [Feb 22 17:23:53] Scheduling destruction of SIP dialog '2731658924@192.168.1.91' in 32000 ms (Method: REGISTER) [Feb 22 17:23:53] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 550 [Feb 22 17:23:53] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 550 [Feb 22 17:23:53] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/550 - state 1 (Not in use) [Feb 22 17:23:53] DEBUG[14811]: app_queue.c:546 changethread: Device 'SIP/550' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Feb 22 17:23:54] <--- 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=z9hG4bK79e6393d From: To: Call-ID: 000ff78d-ebb20007-74946e6d-3266b768@192.168.1.165 Date: Thu, 22 Feb 2007 22:25:28 GMT CSeq: 49726 REGISTER User-Agent: CSCO/7 Contact: Content-Length: 0 Expires: 60 <-------------> [Feb 22 17:23:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:23:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK79e6393d (58) [Feb 22 17:23:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: (40) [Feb 22 17:23:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (38) [Feb 22 17:23:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 000ff78d-ebb20007-74946e6d-3266b768@192.168.1.165 (58) [Feb 22 17:23:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Date: Thu, 22 Feb 2007 22:25:28 GMT (35) [Feb 22 17:23:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: CSeq: 49726 REGISTER (20) [Feb 22 17:23:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: CSCO/7 (18) [Feb 22 17:23:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Contact: (37) [Feb 22 17:23:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:23:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Expires: 60 (11) [Feb 22 17:23:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: (0) [Feb 22 17:23:54] --- (11 headers 0 lines) --- [Feb 22 17:23:54] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 000ff78d-ebb20007-74946e6d-3266b768@192.168.1.165 - REGISTER (No RTP) [Feb 22 17:23:54] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:23:54] Using latest REGISTER request as basis request [Feb 22 17:23:54] Sending to 192.168.1.165 : 5060 (no NAT) [Feb 22 17:23:54] <--- 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=z9hG4bK79e6393d;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20007-74946e6d-3266b768@192.168.1.165 CSeq: 49726 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:23:54] <--- 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=z9hG4bK79e6393d;received=192.168.1.165 From: To: ;tag=as4aeffa89 Call-ID: 000ff78d-ebb20007-74946e6d-3266b768@192.168.1.165 CSeq: 49726 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="505c6855" Content-Length: 0 <------------> [Feb 22 17:23:54] Scheduling destruction of SIP dialog '000ff78d-ebb20007-74946e6d-3266b768@192.168.1.165' in 32000 ms (Method: REGISTER) [Feb 22 17:23:54] <--- 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=z9hG4bK28a53f83 From: To: Call-ID: 000ff78d-ebb20007-74946e6d-3266b768@192.168.1.165 Date: Thu, 22 Feb 2007 22:25:28 GMT CSeq: 49727 REGISTER User-Agent: CSCO/7 Contact: Authorization: Digest username="515",realm="asterisk",uri="sip:24.123.23.170",response="20508e886d96ae4d37830772e4018a1f",nonce="505c6855",algorithm=MD5 Content-Length: 0 Expires: 60 <-------------> [Feb 22 17:23:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:23:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK28a53f83 (58) [Feb 22 17:23:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: (40) [Feb 22 17:23:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (38) [Feb 22 17:23:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 000ff78d-ebb20007-74946e6d-3266b768@192.168.1.165 (58) [Feb 22 17:23:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Date: Thu, 22 Feb 2007 22:25:28 GMT (35) [Feb 22 17:23:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: CSeq: 49727 REGISTER (20) [Feb 22 17:23:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: CSCO/7 (18) [Feb 22 17:23:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Contact: (37) [Feb 22 17:23:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Authorization: Digest username="515",realm="asterisk",uri="sip:24.123.23.170",response="20508e886d96ae4d37830772e4018a1f",nonce="505c6855",algorithm=MD5 (152) [Feb 22 17:23:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Content-Length: 0 (17) [Feb 22 17:23:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Expires: 60 (11) [Feb 22 17:23:54] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: (0) [Feb 22 17:23:54] --- (12 headers 0 lines) --- [Feb 22 17:23:54] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:23:54] Using latest REGISTER request as basis request [Feb 22 17:23:54] Sending to 192.168.1.165 : 5060 (no NAT) [Feb 22 17:23:54] <--- 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=z9hG4bK28a53f83;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20007-74946e6d-3266b768@192.168.1.165 CSeq: 49727 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:23:54] <--- 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=z9hG4bK28a53f83;received=192.168.1.165 From: To: ;tag=as4aeffa89 Call-ID: 000ff78d-ebb20007-74946e6d-3266b768@192.168.1.165 CSeq: 49727 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Thu, 22 Feb 2007 22:23:54 GMT Content-Length: 0 <------------> [Feb 22 17:23:54] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/515 [Feb 22 17:23:54] Scheduling destruction of SIP dialog '000ff78d-ebb20007-74946e6d-3266b768@192.168.1.165' in 32000 ms (Method: REGISTER) [Feb 22 17:23:54] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 515 [Feb 22 17:23:54] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 515 [Feb 22 17:23:54] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/515 - state 1 (Not in use) [Feb 22 17:23:54] DEBUG[14812]: app_queue.c:546 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. [Feb 22 17:23:55] <--- 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=z9hG4bKcfea502cd84ca6c3f49e35b786da625a6 Call-ID: 55ae664696c0@24.123.23.170 CSeq: 58871 REGISTER From: ;tag=ee89218d13667d95921b558002f5cfe0 To: Contact: ;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="41d99ab7", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="641202dafb78240bec6b7287e55fc72a" <-------------> [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.98:5060;branch=z9hG4bKcfea502cd84ca6c3f49e35b786da625a6 (82) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: Call-ID: 55ae664696c0@24.123.23.170 (35) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: CSeq: 58871 REGISTER (20) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: ;tag=ee89218d13667d95921b558002f5cfe0 (66) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: To: (27) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="41d99ab7", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="641202dafb78240bec6b7287e55fc72a" (167) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 13: (0) [Feb 22 17:23:55] --- (13 headers 0 lines) --- [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 55ae664696c0@24.123.23.170 - REGISTER (No RTP) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:23:55] Using latest REGISTER request as basis request [Feb 22 17:23:55] Sending to 192.168.1.98 : 5060 (no NAT) [Feb 22 17:23:55] <--- 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=z9hG4bKcfea502cd84ca6c3f49e35b786da625a6;received=192.168.1.98 From: ;tag=ee89218d13667d95921b558002f5cfe0 To: Call-ID: 55ae664696c0@24.123.23.170 CSeq: 58871 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:23:55] <--- 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=z9hG4bKcfea502cd84ca6c3f49e35b786da625a6;received=192.168.1.98 From: ;tag=ee89218d13667d95921b558002f5cfe0 To: ;tag=as649339c8 Call-ID: 55ae664696c0@24.123.23.170 CSeq: 58871 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="03148311" Content-Length: 0 <------------> [Feb 22 17:23:55] Scheduling destruction of SIP dialog '55ae664696c0@24.123.23.170' in 32000 ms (Method: REGISTER) [Feb 22 17:23:55] <--- 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=z9hG4bKa208e6d42f3794dd4f51659a720adb449 CSeq: 58872 REGISTER Call-ID: 55ae664696c0@24.123.23.170 From: ;tag=ee89218d13667d95921b558002f5cfe0 To: Contact: ;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="03148311", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="77c5afe31f1f0426d1404601b669358f" <-------------> [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.98:5060;branch=z9hG4bKa208e6d42f3794dd4f51659a720adb449 (82) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: CSeq: 58872 REGISTER (20) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: Call-ID: 55ae664696c0@24.123.23.170 (35) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: ;tag=ee89218d13667d95921b558002f5cfe0 (66) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: To: (27) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="03148311", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="77c5afe31f1f0426d1404601b669358f" (167) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 13: (0) [Feb 22 17:23:55] --- (13 headers 0 lines) --- [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:23:55] Using latest REGISTER request as basis request [Feb 22 17:23:55] Sending to 192.168.1.98 : 5060 (no NAT) [Feb 22 17:23:55] <--- 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=z9hG4bKa208e6d42f3794dd4f51659a720adb449;received=192.168.1.98 From: ;tag=ee89218d13667d95921b558002f5cfe0 To: Call-ID: 55ae664696c0@24.123.23.170 CSeq: 58872 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:23:55] <--- 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=z9hG4bKa208e6d42f3794dd4f51659a720adb449;received=192.168.1.98 From: ;tag=ee89218d13667d95921b558002f5cfe0 To: ;tag=as649339c8 Call-ID: 55ae664696c0@24.123.23.170 CSeq: 58872 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Thu, 22 Feb 2007 22:23:55 GMT Content-Length: 0 <------------> [Feb 22 17:23:55] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/529 [Feb 22 17:23:55] Scheduling destruction of SIP dialog '55ae664696c0@24.123.23.170' in 32000 ms (Method: REGISTER) [Feb 22 17:23:55] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 529 [Feb 22 17:23:55] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 529 [Feb 22 17:23:55] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/529 - state 1 (Not in use) [Feb 22 17:23:55] DEBUG[14813]: app_queue.c:546 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. [Feb 22 17:23:55] NOTICE[14767]: chan_sip.c:7055 sip_reregister: -- Re-registration for 3173241052@sip.broadvoice.com [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 42fcb9a2205a226e1657915a20f1c05a@24.123.23.170 - REGISTER (No RTP) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:7214 transmit_register: Scheduled a registration timeout for sip.broadvoice.com id #165 [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:sip.broadvoice.com SIP/2.0 (39) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK12e752d8;rport (64) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: ;tag=as418b7c58 (56) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (39) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 42fcb9a2205a226e1657915a20f1c05a@24.123.23.170 (55) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: CSeq: 106 REGISTER (18) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: Max-Forwards: 70 (16) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Expires: 120 (12) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Contact: (39) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Event: registration (19) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Content-Length: 0 (17) [Feb 22 17:23:55] REGISTER 12 headers, 0 lines [Feb 22 17:23:55] REGISTER attempt 1 to 3173241052@sip.broadvoice.com [Feb 22 17:23:55] 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=z9hG4bK12e752d8;rport From: ;tag=as418b7c58 To: Call-ID: 42fcb9a2205a226e1657915a20f1c05a@24.123.23.170 CSeq: 106 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: Event: registration Content-Length: 0 --- [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #166 [Feb 22 17:23:55] <--- SIP read from 147.135.12.128:5060 ---> SIP/2.0 200 OK Call-ID: 42fcb9a2205a226e1657915a20f1c05a@24.123.23.170 CSeq: 106 REGISTER From: ;tag=as418b7c58 To: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK12e752d8 Contact: Expires: 30 Event: registration Content-Length: 0 <-------------> [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Call-ID: 42fcb9a2205a226e1657915a20f1c05a@24.123.23.170 (55) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: CSeq: 106 REGISTER (18) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: From: ;tag=as418b7c58 (56) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: To: (39) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK12e752d8 (58) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: (39) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: Expires: 30 (11) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Event: registration (19) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (20) [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: (0) [Feb 22 17:23:55] --- (10 headers 0 lines) --- [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:2061 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #166 [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '42fcb9a2205a226e1657915a20f1c05a@24.123.23.170' of Request 106: Match Not Found [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:11906 handle_response_register: Registration successful [Feb 22 17:23:55] DEBUG[14767]: chan_sip.c:11909 handle_response_register: Cancelling timeout 165 [Feb 22 17:23:55] Scheduling destruction of SIP dialog '42fcb9a2205a226e1657915a20f1c05a@24.123.23.170' in 32000 ms (Method: REGISTER) [Feb 22 17:23:55] NOTICE[14767]: chan_sip.c:11961 handle_response_register: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 23 s) [Feb 22 17:23:56] <--- SIP read from 192.168.1.86:5062 ---> <-------------> [Feb 22 17:23:56] --- (0 headers 0 lines) Nat keepalive --- [Feb 22 17:23:58] DEBUG[14767]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '55ae66469f7f@24.123.23.170' [Feb 22 17:23:58] DEBUG[14767]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 55ae66469f7f@24.123.23.170 [Feb 22 17:23:58] Really destroying SIP dialog '55ae66469f7f@24.123.23.170' Method: REGISTER [Feb 22 17:23:58] <--- 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=z9hG4bK5b481104 From: To: Call-ID: 000ff78d-ebb20003-42dc795d-3e449dd6@192.168.1.165 Date: Thu, 22 Feb 2007 22:25:32 GMT CSeq: 49705 REGISTER User-Agent: CSCO/7 Contact: Content-Length: 0 Expires: 60 <-------------> [Feb 22 17:23:58] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:23:58] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK5b481104 (58) [Feb 22 17:23:58] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: (40) [Feb 22 17:23:58] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (38) [Feb 22 17:23:58] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 000ff78d-ebb20003-42dc795d-3e449dd6@192.168.1.165 (58) [Feb 22 17:23:58] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Date: Thu, 22 Feb 2007 22:25:32 GMT (35) [Feb 22 17:23:58] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: CSeq: 49705 REGISTER (20) [Feb 22 17:23:58] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: CSCO/7 (18) [Feb 22 17:23:58] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Contact: (37) [Feb 22 17:23:58] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:23:58] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Expires: 60 (11) [Feb 22 17:23:58] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: (0) [Feb 22 17:23:58] --- (11 headers 0 lines) --- [Feb 22 17:23:58] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 000ff78d-ebb20003-42dc795d-3e449dd6@192.168.1.165 - REGISTER (No RTP) [Feb 22 17:23:58] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:23:58] Using latest REGISTER request as basis request [Feb 22 17:23:58] Sending to 192.168.1.165 : 5060 (no NAT) [Feb 22 17:23:58] <--- 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=z9hG4bK5b481104;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20003-42dc795d-3e449dd6@192.168.1.165 CSeq: 49705 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:23:58] <--- 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=z9hG4bK5b481104;received=192.168.1.165 From: To: ;tag=as2780bfe4 Call-ID: 000ff78d-ebb20003-42dc795d-3e449dd6@192.168.1.165 CSeq: 49705 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="77cbae8a" Content-Length: 0 <------------> [Feb 22 17:23:58] Scheduling destruction of SIP dialog '000ff78d-ebb20003-42dc795d-3e449dd6@192.168.1.165' in 32000 ms (Method: REGISTER) [Feb 22 17:23:59] <--- 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=z9hG4bK5f3cdc39 From: To: Call-ID: 000ff78d-ebb20003-42dc795d-3e449dd6@192.168.1.165 Date: Thu, 22 Feb 2007 22:25:32 GMT CSeq: 49706 REGISTER User-Agent: CSCO/7 Contact: Authorization: Digest username="511",realm="asterisk",uri="sip:24.123.23.170",response="201e5c76e36e8e2a4a44cfd706d3b99e",nonce="77cbae8a",algorithm=MD5 Content-Length: 0 Expires: 60 <-------------> [Feb 22 17:23:59] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:23:59] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK5f3cdc39 (58) [Feb 22 17:23:59] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: (40) [Feb 22 17:23:59] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (38) [Feb 22 17:23:59] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 000ff78d-ebb20003-42dc795d-3e449dd6@192.168.1.165 (58) [Feb 22 17:23:59] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Date: Thu, 22 Feb 2007 22:25:32 GMT (35) [Feb 22 17:23:59] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: CSeq: 49706 REGISTER (20) [Feb 22 17:23:59] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: CSCO/7 (18) [Feb 22 17:23:59] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Contact: (37) [Feb 22 17:23:59] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Authorization: Digest username="511",realm="asterisk",uri="sip:24.123.23.170",response="201e5c76e36e8e2a4a44cfd706d3b99e",nonce="77cbae8a",algorithm=MD5 (152) [Feb 22 17:23:59] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Content-Length: 0 (17) [Feb 22 17:23:59] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Expires: 60 (11) [Feb 22 17:23:59] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: (0) [Feb 22 17:23:59] --- (12 headers 0 lines) --- [Feb 22 17:23:59] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:23:59] Using latest REGISTER request as basis request [Feb 22 17:23:59] Sending to 192.168.1.165 : 5060 (no NAT) [Feb 22 17:23:59] <--- 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=z9hG4bK5f3cdc39;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20003-42dc795d-3e449dd6@192.168.1.165 CSeq: 49706 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:23:59] <--- 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=z9hG4bK5f3cdc39;received=192.168.1.165 From: To: ;tag=as2780bfe4 Call-ID: 000ff78d-ebb20003-42dc795d-3e449dd6@192.168.1.165 CSeq: 49706 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Thu, 22 Feb 2007 22:23:59 GMT Content-Length: 0 <------------> [Feb 22 17:23:59] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/511 [Feb 22 17:23:59] Scheduling destruction of SIP dialog '000ff78d-ebb20003-42dc795d-3e449dd6@192.168.1.165' in 32000 ms (Method: REGISTER) [Feb 22 17:23:59] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 511 [Feb 22 17:23:59] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 511 [Feb 22 17:23:59] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/511 - state 1 (Not in use) [Feb 22 17:23:59] DEBUG[14814]: app_queue.c:546 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. [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '55ae66476793@24.123.23.170' [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 55ae66476793@24.123.23.170 [Feb 22 17:24:00] Really destroying SIP dialog '55ae66476793@24.123.23.170' Method: REGISTER [Feb 22 17:24:00] <--- SIP read from 192.168.1.66:5060 ---> INVITE sip:204@24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bKwe0667f489c0daa395f97a9cf75e8dfd4 Call-ID: 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 CSeq: 467579 INVITE From: 526 ;tag=48106fce45713fff4b942431e90afa0e To: Contact: Session-Expires: 300 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=- 1783940127 163215 IN IP4 192.168.1.66 s=- c=IN IP4 192.168.1.66 t=0 0 m=audio 30006 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 <-------------> [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: INVITE sip:204@24.123.23.170 SIP/2.0 (36) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bKwe0667f489c0daa395f97a9cf75e8dfd4 (82) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: Call-ID: 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 (68) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: CSeq: 467579 INVITE (19) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: 526 ;tag=48106fce45713fff4b942431e90afa0e (70) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: To: (27) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: (36) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: Session-Expires: 300 (20) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Content-Type: application/sdp (29) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Max-Forwards: 70 (16) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 13: Content-Length: 269 (19) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 14: (0) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: v=0 (3) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: o=- 1783940127 163215 IN IP4 192.168.1.66 (41) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: s=- (3) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: c=IN IP4 192.168.1.66 (21) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: t=0 0 (5) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: m=audio 30006 RTP/AVP 0 8 18 101 (32) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=fmtp:101 0-15,16,17,18 (24) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=sendrecv (10) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=ptime:20 (10) [Feb 22 17:24:00] --- (14 headers 13 lines) --- [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:2559 do_setnat: Setting NAT on RTP to Off [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:2564 do_setnat: Setting NAT on VRTP to Off [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 - INVITE (With RTP) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:1662 parse_sip_options: Begin: parsing SIP "Supported: sip-cc, sip-cc-01, replaces, timer" [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:1670 parse_sip_options: Found SIP option: -sip-cc- [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:1684 parse_sip_options: Found no match for SIP option: sip-cc (Please file bug report!) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:1670 parse_sip_options: Found SIP option: -sip-cc-01- [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:1684 parse_sip_options: Found no match for SIP option: sip-cc-01 (Please file bug report!) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:1670 parse_sip_options: Found SIP option: -replaces- [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:1676 parse_sip_options: Matched SIP option: replaces [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:1670 parse_sip_options: Found SIP option: -timer- [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:1676 parse_sip_options: Matched SIP option: timer [Feb 22 17:24:00] Sending to 192.168.1.66 : 5060 (no NAT) [Feb 22 17:24:00] Using INVITE request as basis request - 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:2559 do_setnat: Setting NAT on RTP to Off [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:2564 do_setnat: Setting NAT on VRTP to Off [Feb 22 17:24:00] <--- Reliably Transmitting (no NAT) to 192.168.1.66:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bKwe0667f489c0daa395f97a9cf75e8dfd4;received=192.168.1.66 From: 526 ;tag=48106fce45713fff4b942431e90afa0e To: ;tag=as29a78c3e Call-ID: 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 CSeq: 467579 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="557b5136" Content-Length: 0 <------------> [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #172 [Feb 22 17:24:00] Scheduling destruction of SIP dialog '903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170' in 32000 ms (Method: INVITE) [Feb 22 17:24:00] Found user '526' [Feb 22 17:24:00] <--- SIP read from 192.168.1.66:5060 ---> ACK sip:204@24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bKwe0667f489c0daa395f97a9cf75e8dfd4 CSeq: 467579 ACK To: ;tag=as29a78c3e Call-ID: 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 From: 526 ;tag=48106fce45713fff4b942431e90afa0e User-Agent: Uniden SIP Phone p2 Ver BS4.77 <-------------> [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: ACK sip:204@24.123.23.170 SIP/2.0 (33) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bKwe0667f489c0daa395f97a9cf75e8dfd4 (82) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: CSeq: 467579 ACK (16) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: ;tag=as29a78c3e (42) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 (68) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: From: 526 ;tag=48106fce45713fff4b942431e90afa0e (70) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: (0) [Feb 22 17:24:00] --- (7 headers 0 lines) --- [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received ACK (6) - Command in SIP ACK [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:2061 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #172 [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170' of Response 467579: Match Not Found [Feb 22 17:24:00] <--- SIP read from 192.168.1.66:5060 ---> INVITE sip:204@24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bKs5c49dfabb5fb924d0c5cf2cee19845ff CSeq: 467580 INVITE Call-ID: 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 From: 526 ;tag=48106fce45713fff4b942431e90afa0e To: Contact: Session-Expires: 300 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 Proxy-Authorization: Digest realm="asterisk", nonce="557b5136", algorithm=MD5, uri="sip:204@24.123.23.170", username="526", response="201dd03b55dbc67b903bc1ef09fcc799" v=0 o=- 1783940127 163215 IN IP4 192.168.1.66 s=- c=IN IP4 192.168.1.66 t=0 0 m=audio 30006 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 <-------------> [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: INVITE sip:204@24.123.23.170 SIP/2.0 (36) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bKs5c49dfabb5fb924d0c5cf2cee19845ff (82) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: CSeq: 467580 INVITE (19) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: Call-ID: 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 (68) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: 526 ;tag=48106fce45713fff4b942431e90afa0e (70) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: To: (27) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: (36) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: Session-Expires: 300 (20) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Content-Type: application/sdp (29) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Max-Forwards: 70 (16) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 13: Content-Length: 269 (19) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 14: Proxy-Authorization: Digest realm="asterisk", nonce="557b5136", algorithm=MD5, uri="sip:204@24.123.23.170", username="526", response="201dd03b55dbc67b903bc1ef09fcc799" (177) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 15: (0) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: v=0 (3) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: o=- 1783940127 163215 IN IP4 192.168.1.66 (41) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: s=- (3) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: c=IN IP4 192.168.1.66 (21) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: t=0 0 (5) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: m=audio 30006 RTP/AVP 0 8 18 101 (32) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=fmtp:101 0-15,16,17,18 (24) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=sendrecv (10) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=ptime:20 (10) [Feb 22 17:24:00] --- (15 headers 13 lines) --- [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Feb 22 17:24:00] Sending to 192.168.1.66 : 5060 (no NAT) [Feb 22 17:24:00] Using INVITE request as basis request - 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:2559 do_setnat: Setting NAT on RTP to Off [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:2564 do_setnat: Setting NAT on VRTP to Off [Feb 22 17:24:00] Found user '526' [Feb 22 17:24:00] Found RTP audio format 0 [Feb 22 17:24:00] Found RTP audio format 8 [Feb 22 17:24:00] Found RTP audio format 18 [Feb 22 17:24:00] Found RTP audio format 101 [Feb 22 17:24:00] Peer audio RTP is at port 192.168.1.66:30006 [Feb 22 17:24:00] Found description format PCMU for ID 0 [Feb 22 17:24:00] Found description format PCMA for ID 8 [Feb 22 17:24:00] Found description format G729 for ID 18 [Feb 22 17:24:00] Found description format telephone-event for ID 101 [Feb 22 17:24:00] Got unsupported a:fmtp in SDP offer [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:5068 process_sdp: T38 state changed to 0 on channel [Feb 22 17:24:00] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) [Feb 22 17:24:00] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Feb 22 17:24:00] Peer audio RTP is at port 192.168.1.66:30006 [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:5145 process_sdp: We're settling with these formats: 0xc (ulaw|alaw) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:13237 handle_request_invite: Checking SIP call limits for device 526 [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:2978 update_call_counter: Updating call counter for incoming call [Feb 22 17:24:00] Looking for 204 in smvoice-sip (domain 24.123.23.170) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:3766 sip_new: *** Our native formats are 0x4 (ulaw) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:3767 sip_new: *** Joint capabilities are 0xc (ulaw|alaw) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:3768 sip_new: *** Our capabilities are 0xe (gsm|ulaw|alaw) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:3769 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:3792 sip_new: This channel will not be able to handle video. [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:7891 build_route: build_route: Contact hop: [Feb 22 17:24:00] list_route: hop: [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:13310 handle_request_invite: SIP/526-09df30e0: New call is still down.... Trying... [Feb 22 17:24:00] <--- 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=z9hG4bKs5c49dfabb5fb924d0c5cf2cee19845ff;received=192.168.1.66 From: 526 ;tag=48106fce45713fff4b942431e90afa0e To: Call-ID: 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 CSeq: 467580 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:24:00] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/526-09df30e0 [Feb 22 17:24:00] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 526 [Feb 22 17:24:00] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 526 [Feb 22 17:24:00] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/526 - state 1 (Not in use) [Feb 22 17:24:00] DEBUG[14821]: pbx.c:1767 pbx_extension_helper: Launching 'NoOp' [Feb 22 17:24:00] -- Executing [204@smvoice-sip:1] NoOp("SIP/526-09df30e0", "2xx") in new stack [Feb 22 17:24:00] DEBUG[14821]: pbx.c:1767 pbx_extension_helper: Launching 'Set' [Feb 22 17:24:00] -- Executing [204@smvoice-sip:2] Set("SIP/526-09df30e0", "SMVOICE_CONTEXT_EXTEN=204") in new stack [Feb 22 17:24:00] DEBUG[14821]: pbx.c:1767 pbx_extension_helper: Launching 'AGI' [Feb 22 17:24:00] -- Executing [204@smvoice-sip:3] AGI("SIP/526-09df30e0", "smvoice|-company|demonstration|-digium_asterisk|-asterisk_callat_forwarding|204") in new stack [Feb 22 17:24:00] -- Launched AGI Script /var/lib/asterisk/agi-bin/smvoice [Feb 22 17:24:00] DEBUG[14822]: app_queue.c:546 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. [Feb 22 17:24:00] -- AGI Script smvoice completed, returning 0 [Feb 22 17:24:00] DEBUG[14821]: pbx.c:1688 pbx_substitute_variables_helper_full: Expression result is '0' [Feb 22 17:24:00] DEBUG[14821]: pbx.c:1767 pbx_extension_helper: Launching 'GotoIf' [Feb 22 17:24:00] -- Executing [204@smvoice-sip:4] GotoIf("SIP/526-09df30e0", "0?INVALID|1") in new stack [Feb 22 17:24:00] DEBUG[14821]: pbx.c:5932 pbx_builtin_gotoif: Not taking any branch [Feb 22 17:24:00] DEBUG[14821]: pbx.c:1688 pbx_substitute_variables_helper_full: Expression result is '0' [Feb 22 17:24:00] DEBUG[14821]: pbx.c:1767 pbx_extension_helper: Launching 'GotoIf' [Feb 22 17:24:00] -- Executing [204@smvoice-sip:5] GotoIf("SIP/526-09df30e0", "0?_2XX-NOANSWER|1") in new stack [Feb 22 17:24:00] DEBUG[14821]: pbx.c:5932 pbx_builtin_gotoif: Not taking any branch [Feb 22 17:24:00] DEBUG[14821]: pbx.c:1767 pbx_extension_helper: Launching 'Dial' [Feb 22 17:24:00] -- Executing [204@smvoice-sip:6] Dial("SIP/526-09df30e0", "SIP/528|20|") in new stack [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:15106 sip_request_call: Asked to create a SIP channel with formats: 0x4 (ulaw) [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:2559 do_setnat: Setting NAT on RTP to Off [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:3766 sip_new: *** Our native formats are 0x4 (ulaw) [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:3767 sip_new: *** Joint capabilities are 0x0 (nothing) [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:3768 sip_new: *** Our capabilities are 0xe (gsm|ulaw|alaw) [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:3769 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:3771 sip_new: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:3792 sip_new: This channel will not be able to handle video. [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:16701 sip_set_rtp_peer: Early remote bridge setting SIP '524091ef199964d13508e0184586dc1c@24.123.23.170' - Sending media to 192.168.1.66 [Feb 22 17:24:00] DEBUG[14821]: rtp.c:1570 ast_rtp_make_compatible: Seeded SDP of 'SIP/528-09e02620' with that of 'SIP/526-09df30e0' [Feb 22 17:24:00] DEBUG[14821]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-smvoice-sip-204-6. [Feb 22 17:24:00] DEBUG[14821]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-smvoice-sip-204-5. [Feb 22 17:24:00] DEBUG[14821]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-smvoice-sip-204-4. [Feb 22 17:24:00] DEBUG[14821]: channel.c:3185 ast_channel_inherit_variables: Not copying variable AGISTATUS. [Feb 22 17:24:00] DEBUG[14821]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SMVOICE_WORK_EXTENSION. [Feb 22 17:24:00] DEBUG[14821]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SMVOICE_EXTENSION. [Feb 22 17:24:00] DEBUG[14821]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SMVOICE_PIN. [Feb 22 17:24:00] DEBUG[14821]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SMVOICE_CALLAT. [Feb 22 17:24:00] DEBUG[14821]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-smvoice-sip-204-3. [Feb 22 17:24:00] DEBUG[14821]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SMVOICE_CONTEXT_EXTEN. [Feb 22 17:24:00] DEBUG[14821]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-smvoice-sip-204-2. [Feb 22 17:24:00] DEBUG[14821]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-smvoice-sip-204-1. [Feb 22 17:24:00] DEBUG[14821]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Feb 22 17:24:00] DEBUG[14821]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT. [Feb 22 17:24:00] DEBUG[14821]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Feb 22 17:24:00] DEBUG[14821]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPURI. [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:2812 sip_call: Outgoing Call for 528 [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:2978 update_call_counter: Updating call counter for outgoing call [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:2826 sip_call: Our T38 capability (0), joint T38 capability (0) [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:6122 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: False [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:6123 add_sdp: ** Our prefcodec: 0x4 (ulaw) [Feb 22 17:24:00] Audio is at 24.123.23.170 port 16212 [Feb 22 17:24:00] Adding codec 0x4 (ulaw) to SDP [Feb 22 17:24:00] Adding codec 0x8 (alaw) to SDP [Feb 22 17:24:00] Adding codec 0x2 (gsm) to SDP [Feb 22 17:24:00] Adding non-codec 0x1 (telephone-event) to SDP [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:6254 add_sdp: -- Done with adding codecs to SDP [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:6299 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 0: INVITE sip:528@192.168.1.93:5060 SIP/2.0 (40) [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK1b6f1d48 (58) [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 2: From: "526 526" ;tag=as449238ce (54) [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 3: To: (31) [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 4: Contact: (32) [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 5: Call-ID: 524091ef199964d13508e0184586dc1c@24.123.23.170 (55) [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 6: CSeq: 102 INVITE (16) [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 9: Date: Thu, 22 Feb 2007 22:24:00 GMT (35) [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 11: Supported: replaces (19) [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 12: Content-Type: application/sdp (29) [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 13: Content-Length: 287 (19) [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 14: (0) [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: v=0 (3) [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: o=root 14736 14736 IN IP4 192.168.1.66 (38) [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: s=session (9) [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: c=IN IP4 192.168.1.66 (21) [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: t=0 0 (5) [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: m=audio 30006 RTP/AVP 0 8 3 101 (31) [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: a=fmtp:101 0-16 (15) [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: a=silenceSupp:off - - - - (25) [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: a=ptime:20 (10) [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: a=sendrecv (10) [Feb 22 17:24:00] Reliably Transmitting (no NAT) to 192.168.1.93:5060: INVITE sip:528@192.168.1.93:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK1b6f1d48 From: "526 526" ;tag=as449238ce To: Contact: Call-ID: 524091ef199964d13508e0184586dc1c@24.123.23.170 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 22 Feb 2007 22:24:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 287 v=0 o=root 14736 14736 IN IP4 192.168.1.66 s=session c=IN IP4 192.168.1.66 t=0 0 m=audio 30006 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Feb 22 17:24:00] DEBUG[14821]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #174 [Feb 22 17:24:00] -- Called 528 [Feb 22 17:24:00] <--- SIP read from 192.168.1.93:5060 ---> SIP/2.0 100 Trying To: ;tag=81c5c0cdcd26df8f2948fc3e1971f0eb Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK1b6f1d48 From: "526 526" ;tag=as449238ce Call-ID: 524091ef199964d13508e0184586dc1c@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 <-------------> [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 100 Trying (18) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: To: ;tag=81c5c0cdcd26df8f2948fc3e1971f0eb (68) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK1b6f1d48 (58) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: From: "526 526" ;tag=as449238ce (54) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 524091ef199964d13508e0184586dc1c@24.123.23.170 (55) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: CSeq: 102 INVITE (16) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Content-Length: 0 (17) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: (0) [Feb 22 17:24:00] --- (9 headers 0 lines) --- [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:2104 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #174 - INVITE (got response) [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:2113 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '524091ef199964d13508e0184586dc1c@24.123.23.170' Request 102: Found [Feb 22 17:24:00] DEBUG[14767]: chan_sip.c:11531 handle_response_invite: SIP response 100 to standard invite [Feb 22 17:24:01] <--- SIP read from 192.168.1.93:5060 ---> SIP/2.0 180 Ringing To: ;tag=81c5c0cdcd26df8f2948fc3e1971f0eb Contact: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK1b6f1d48 From: "526 526" ;tag=as449238ce Call-ID: 524091ef199964d13508e0184586dc1c@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 <-------------> [Feb 22 17:24:01] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 180 Ringing (19) [Feb 22 17:24:01] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: To: ;tag=81c5c0cdcd26df8f2948fc3e1971f0eb (68) [Feb 22 17:24:01] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: Contact: (36) [Feb 22 17:24:01] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK1b6f1d48 (58) [Feb 22 17:24:01] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: "526 526" ;tag=as449238ce (54) [Feb 22 17:24:01] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Call-ID: 524091ef199964d13508e0184586dc1c@24.123.23.170 (55) [Feb 22 17:24:01] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: CSeq: 102 INVITE (16) [Feb 22 17:24:01] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: Content-Length: 0 (17) [Feb 22 17:24:01] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:24:01] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:24:01] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: (0) [Feb 22 17:24:01] --- (10 headers 0 lines) --- [Feb 22 17:24:01] DEBUG[14767]: chan_sip.c:2113 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '524091ef199964d13508e0184586dc1c@24.123.23.170' Request 102: Found [Feb 22 17:24:01] DEBUG[14767]: chan_sip.c:11531 handle_response_invite: SIP response 180 to standard invite [Feb 22 17:24:01] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/528-09e02620 [Feb 22 17:24:01] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 528 [Feb 22 17:24:01] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 528 [Feb 22 17:24:01] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/528 - state 1 (Not in use) [Feb 22 17:24:01] DEBUG[14824]: app_queue.c:546 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. [Feb 22 17:24:01] -- SIP/528-09e02620 is ringing [Feb 22 17:24:01] DEBUG[14821]: rtp.c:1505 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/526-09df30e0' with that of 'SIP/528-09e02620' [Feb 22 17:24:01] <--- Transmitting (no NAT) to 192.168.1.66:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bKs5c49dfabb5fb924d0c5cf2cee19845ff;received=192.168.1.66 From: 526 ;tag=48106fce45713fff4b942431e90afa0e To: ;tag=as056fdf38 Call-ID: 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 CSeq: 467580 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:24:03] <--- SIP read from 192.168.1.93:5060 ---> SIP/2.0 200 OK To: ;tag=81c5c0cdcd26df8f2948fc3e1971f0eb Contact: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK1b6f1d48 From: "526 526" ;tag=as449238ce Call-ID: 524091ef199964d13508e0184586dc1c@24.123.23.170 CSeq: 102 INVITE Content-Type: application/sdp Content-Length: 246 Supported: sip-cc, sip-cc-01, replaces, timer Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK v=0 o=- 14736 70056 IN IP4 192.168.1.93 s=session c=IN IP4 192.168.1.93 t=0 0 m=audio 30030 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 - - - - a=ptime:20 a=sendrecv <-------------> [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: To: ;tag=81c5c0cdcd26df8f2948fc3e1971f0eb (68) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: Contact: (36) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK1b6f1d48 (58) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: "526 526" ;tag=as449238ce (54) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Call-ID: 524091ef199964d13508e0184586dc1c@24.123.23.170 (55) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: CSeq: 102 INVITE (16) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: Content-Type: application/sdp (29) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Content-Length: 246 (19) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: (0) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: v=0 (3) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: o=- 14736 70056 IN IP4 192.168.1.93 (35) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: s=session (9) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: c=IN IP4 192.168.1.93 (21) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: t=0 0 (5) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: m=audio 30030 RTP/AVP 0 101 (27) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=fmtp:101 0-15,16,17,18 (24) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=silenceSupp:off - - - - (25) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=ptime:20 (10) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=sendrecv (10) [Feb 22 17:24:03] --- (11 headers 12 lines) --- [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:2053 __sip_ack: Acked pending invite 102 [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '524091ef199964d13508e0184586dc1c@24.123.23.170' of Request 102: Match Not Found [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:11531 handle_response_invite: SIP response 200 to standard invite [Feb 22 17:24:03] Found RTP audio format 0 [Feb 22 17:24:03] Found RTP audio format 101 [Feb 22 17:24:03] Peer audio RTP is at port 192.168.1.93:30030 [Feb 22 17:24:03] Found description format PCMU for ID 0 [Feb 22 17:24:03] Found description format telephone-event for ID 101 [Feb 22 17:24:03] Got unsupported a:fmtp in SDP offer [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:5068 process_sdp: T38 state changed to 0 on channel SIP/528-09e02620 [Feb 22 17:24:03] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Feb 22 17:24:03] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Feb 22 17:24:03] Peer audio RTP is at port 192.168.1.93:30030 [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:5145 process_sdp: We're settling with these formats: 0x4 (ulaw) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:5152 process_sdp: We have an owner, now see if we need to change this call [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:2978 update_call_counter: Updating call counter for outgoing call [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:7891 build_route: build_route: Contact hop: [Feb 22 17:24:03] list_route: hop: [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:5576 reqprep: Strict routing enforced for session 524091ef199964d13508e0184586dc1c@24.123.23.170 [Feb 22 17:24:03] set_destination: Parsing for address/port to send to [Feb 22 17:24:03] set_destination: set destination to 192.168.1.93, port 5060 [Feb 22 17:24:03] Transmitting (no NAT) to 192.168.1.93:5060: ACK sip:528@192.168.1.93:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK3f6334fb From: "526 526" ;tag=as449238ce To: ;tag=81c5c0cdcd26df8f2948fc3e1971f0eb Contact: Call-ID: 524091ef199964d13508e0184586dc1c@24.123.23.170 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Feb 22 17:24:03] DEBUG[14821]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/528-09e02620 [Feb 22 17:24:03] -- SIP/528-09e02620 answered SIP/526-09df30e0 [Feb 22 17:24:03] DEBUG[14821]: chan_sip.c:16701 sip_set_rtp_peer: Early remote bridge setting SIP '903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170' - Sending media to 192.168.1.93 [Feb 22 17:24:03] DEBUG[14821]: rtp.c:1505 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/526-09df30e0' with that of 'SIP/528-09e02620' [Feb 22 17:24:03] DEBUG[14821]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/526-09df30e0 [Feb 22 17:24:03] DEBUG[14821]: chan_sip.c:3428 sip_answer: SIP answering channel: SIP/526-09df30e0 [Feb 22 17:24:03] DEBUG[14821]: chan_sip.c:6354 transmit_response_with_sdp: Setting framing from config on incoming call [Feb 22 17:24:03] DEBUG[14821]: chan_sip.c:6122 add_sdp: ** Our capability: 0xc (ulaw|alaw) Video flag: True [Feb 22 17:24:03] DEBUG[14821]: chan_sip.c:6123 add_sdp: ** Our prefcodec: 0x0 (nothing) [Feb 22 17:24:03] Audio is at 24.123.23.170 port 15904 [Feb 22 17:24:03] Adding codec 0x4 (ulaw) to SDP [Feb 22 17:24:03] Adding codec 0x8 (alaw) to SDP [Feb 22 17:24:03] Adding non-codec 0x1 (telephone-event) to SDP [Feb 22 17:24:03] DEBUG[14821]: chan_sip.c:6254 add_sdp: -- Done with adding codecs to SDP [Feb 22 17:24:03] DEBUG[14821]: chan_sip.c:6299 add_sdp: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Feb 22 17:24:03] <--- Reliably 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=z9hG4bKs5c49dfabb5fb924d0c5cf2cee19845ff;received=192.168.1.66 From: 526 ;tag=48106fce45713fff4b942431e90afa0e To: ;tag=as056fdf38 Call-ID: 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 CSeq: 467580 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 264 v=0 o=root 14736 14736 IN IP4 192.168.1.93 s=session c=IN IP4 192.168.1.93 t=0 0 m=audio 30030 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 - - - - a=ptime:20 a=sendrecv <------------> [Feb 22 17:24:03] DEBUG[14821]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #176 [Feb 22 17:24:03] -- Native bridging SIP/526-09df30e0 and SIP/528-09e02620 [Feb 22 17:24:03] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 528 [Feb 22 17:24:03] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 528 [Feb 22 17:24:03] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/528 - state 1 (Not in use) [Feb 22 17:24:03] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 526 [Feb 22 17:24:03] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 526 [Feb 22 17:24:03] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/526 - state 1 (Not in use) [Feb 22 17:24:03] DEBUG[14825]: app_queue.c:546 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. [Feb 22 17:24:03] DEBUG[14826]: app_queue.c:546 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. [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '42fcb9a2205a226e1657915a20f1c05a@24.123.23.170' [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 42fcb9a2205a226e1657915a20f1c05a@24.123.23.170 [Feb 22 17:24:03] Really destroying SIP dialog '42fcb9a2205a226e1657915a20f1c05a@24.123.23.170' Method: REGISTER [Feb 22 17:24:03] <--- SIP read from 192.168.1.66:5060 ---> ACK sip:204@24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bKk7e10b4135da8f7e8a9caaf1658403b26 CSeq: 467580 ACK To: ;tag=as056fdf38 Call-ID: 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 From: 526 ;tag=48106fce45713fff4b942431e90afa0e User-Agent: Uniden SIP Phone p2 Ver BS4.77 Proxy-Authorization: Digest realm="asterisk", nonce="557b5136", algorithm=MD5, uri="sip:204@24.123.23.170", username="526", response="201dd03b55dbc67b903bc1ef09fcc799" <-------------> [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: ACK sip:204@24.123.23.170 SIP/2.0 (33) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bKk7e10b4135da8f7e8a9caaf1658403b26 (82) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: CSeq: 467580 ACK (16) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: ;tag=as056fdf38 (42) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 (68) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: From: 526 ;tag=48106fce45713fff4b942431e90afa0e (70) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: Proxy-Authorization: Digest realm="asterisk", nonce="557b5136", algorithm=MD5, uri="sip:204@24.123.23.170", username="526", response="201dd03b55dbc67b903bc1ef09fcc799" (177) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: (0) [Feb 22 17:24:03] --- (8 headers 0 lines) --- [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received ACK (6) - Command in SIP ACK [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:2061 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #176 [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170' of Response 467580: Match Not Found [Feb 22 17:24:03] <--- 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=z9hG4bK1602461a From: To: Call-ID: 000ff78d-ebb20005-4fe621ee-067da175@192.168.1.165 Date: Thu, 22 Feb 2007 22:25:37 GMT CSeq: 49747 REGISTER User-Agent: CSCO/7 Contact: Content-Length: 0 Expires: 60 <-------------> [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK1602461a (58) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: (40) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (38) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 000ff78d-ebb20005-4fe621ee-067da175@192.168.1.165 (58) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Date: Thu, 22 Feb 2007 22:25:37 GMT (35) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: CSeq: 49747 REGISTER (20) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: CSCO/7 (18) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Contact: (37) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Expires: 60 (11) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: (0) [Feb 22 17:24:03] --- (11 headers 0 lines) --- [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 000ff78d-ebb20005-4fe621ee-067da175@192.168.1.165 - REGISTER (No RTP) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:24:03] Using latest REGISTER request as basis request [Feb 22 17:24:03] Sending to 192.168.1.165 : 5060 (no NAT) [Feb 22 17:24:03] <--- 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=z9hG4bK1602461a;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20005-4fe621ee-067da175@192.168.1.165 CSeq: 49747 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:24:03] <--- 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=z9hG4bK1602461a;received=192.168.1.165 From: To: ;tag=as03fbed91 Call-ID: 000ff78d-ebb20005-4fe621ee-067da175@192.168.1.165 CSeq: 49747 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6f77497d" Content-Length: 0 <------------> [Feb 22 17:24:03] Scheduling destruction of SIP dialog '000ff78d-ebb20005-4fe621ee-067da175@192.168.1.165' in 32000 ms (Method: REGISTER) [Feb 22 17:24:03] <--- 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=z9hG4bK1155711d From: To: Call-ID: 000ff78d-ebb20005-4fe621ee-067da175@192.168.1.165 Date: Thu, 22 Feb 2007 22:25:37 GMT CSeq: 49748 REGISTER User-Agent: CSCO/7 Contact: Authorization: Digest username="513",realm="asterisk",uri="sip:24.123.23.170",response="eef48f330e92197be468bd2744d3369d",nonce="6f77497d",algorithm=MD5 Content-Length: 0 Expires: 60 <-------------> [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK1155711d (58) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: (40) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (38) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 000ff78d-ebb20005-4fe621ee-067da175@192.168.1.165 (58) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Date: Thu, 22 Feb 2007 22:25:37 GMT (35) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: CSeq: 49748 REGISTER (20) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: CSCO/7 (18) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Contact: (37) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Authorization: Digest username="513",realm="asterisk",uri="sip:24.123.23.170",response="eef48f330e92197be468bd2744d3369d",nonce="6f77497d",algorithm=MD5 (152) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Content-Length: 0 (17) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Expires: 60 (11) [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: (0) [Feb 22 17:24:03] --- (12 headers 0 lines) --- [Feb 22 17:24:03] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:24:03] Using latest REGISTER request as basis request [Feb 22 17:24:03] Sending to 192.168.1.165 : 5060 (no NAT) [Feb 22 17:24:03] <--- 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=z9hG4bK1155711d;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20005-4fe621ee-067da175@192.168.1.165 CSeq: 49748 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:24:03] <--- 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=z9hG4bK1155711d;received=192.168.1.165 From: To: ;tag=as03fbed91 Call-ID: 000ff78d-ebb20005-4fe621ee-067da175@192.168.1.165 CSeq: 49748 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Thu, 22 Feb 2007 22:24:03 GMT Content-Length: 0 <------------> [Feb 22 17:24:03] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/513 [Feb 22 17:24:03] Scheduling destruction of SIP dialog '000ff78d-ebb20005-4fe621ee-067da175@192.168.1.165' in 32000 ms (Method: REGISTER) [Feb 22 17:24:03] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 513 [Feb 22 17:24:03] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 513 [Feb 22 17:24:03] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/513 - state 1 (Not in use) [Feb 22 17:24:03] DEBUG[14827]: app_queue.c:546 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. [Feb 22 17:24:06] <--- SIP read from 192.168.1.86:5060 ---> <-------------> [Feb 22 17:24:06] --- (0 headers 0 lines) Nat keepalive --- [Feb 22 17:24:08] <--- SIP read from 68.58.36.157:5060 ---> <-------------> [Feb 22 17:24:08] --- (0 headers 0 lines) Nat keepalive --- [Feb 22 17:24:09] <--- SIP read from 192.168.1.91:5060 ---> INVITE sip:204@24.123.23.170:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.91:5060;rport;branch=z9hG4bK1478431787 From: "Display Name" ;tag=2772323005 To: Call-ID: 1735147470@192.168.1.91 CSeq: 1000 INVITE Contact: max-forwards: 70 user-agent: UTSTARCOM F3000/Device ID-F3000_TEST Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, MESSAGE, REGISTER, REFER, PING, NOTIFY, INFO, PRACK Content-Type: application/sdp Content-Length: 298 v=0 o=550 123456 654322 IN IP4 192.168.1.91 s=none c=IN IP4 192.168.1.91 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 <-------------> [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: INVITE sip:204@24.123.23.170:5060 SIP/2.0 (41) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.91:5060;rport;branch=z9hG4bK1478431787 (65) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: "Display Name" ;tag=2772323005 (64) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (32) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 1735147470@192.168.1.91 (32) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: CSeq: 1000 INVITE (17) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: (36) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: max-forwards: 70 (16) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: user-agent: UTSTARCOM F3000/Device ID-F3000_TEST (48) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, MESSAGE, REGISTER, REFER, PING, NOTIFY, INFO, PRACK (93) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Content-Type: application/sdp (29) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Content-Length: 298 (21) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: (0) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: v=0 (3) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: o=550 123456 654322 IN IP4 192.168.1.91 (39) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: s=none (6) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: c=IN IP4 192.168.1.91 (21) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: t=0 0 (5) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: m=audio 10010 RTP/AVP 0 8 18 2 101 (34) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=ptime:20 (10) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:18 G729A/8000 (22) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=fmtp:18 annexb=no (19) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:2 G726-32/8000 (23) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=fmtp:101 0-15 (15) [Feb 22 17:24:09] --- (12 headers 14 lines) --- [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:2559 do_setnat: Setting NAT on RTP to Off [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:2564 do_setnat: Setting NAT on VRTP to Off [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 1735147470@192.168.1.91 - INVITE (With RTP) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Feb 22 17:24:09] Sending to 192.168.1.91 : 5060 (NAT) [Feb 22 17:24:09] Using INVITE request as basis request - 1735147470@192.168.1.91 [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:2559 do_setnat: Setting NAT on RTP to On [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:2564 do_setnat: Setting NAT on VRTP to On [Feb 22 17:24:09] <--- Reliably Transmitting (NAT) to 192.168.1.91:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.91:5060;branch=z9hG4bK1478431787;received=192.168.1.91;rport=5060 From: "Display Name" ;tag=2772323005 To: ;tag=as60a681ad Call-ID: 1735147470@192.168.1.91 CSeq: 1000 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0db53aaa" Content-Length: 0 <------------> [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #180 [Feb 22 17:24:09] Scheduling destruction of SIP dialog '1735147470@192.168.1.91' in 32000 ms (Method: INVITE) [Feb 22 17:24:09] Found user '550' [Feb 22 17:24:09] <--- SIP read from 192.168.1.91:5060 ---> ACK sip:204@24.123.23.170:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.91:5060;rport;branch=z9hG4bK1478431787 From: "Display Name" ;tag=2772323005 To: ;tag=as60a681ad Call-ID: 1735147470@192.168.1.91 CSeq: 1000 ACK Content-Length: 0 <-------------> [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: ACK sip:204@24.123.23.170:5060 SIP/2.0 (38) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.91:5060;rport;branch=z9hG4bK1478431787 (65) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: "Display Name" ;tag=2772323005 (64) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: ;tag=as60a681ad (47) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 1735147470@192.168.1.91 (32) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: CSeq: 1000 ACK (14) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Content-Length: 0 (17) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: (0) [Feb 22 17:24:09] --- (7 headers 0 lines) --- [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received ACK (6) - Command in SIP ACK [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:2061 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #180 [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '1735147470@192.168.1.91' of Response 1000: Match Not Found [Feb 22 17:24:09] <--- SIP read from 192.168.1.91:5060 ---> INVITE sip:204@24.123.23.170:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.91:5060;rport;branch=z9hG4bK4050918756 From: "Display Name" ;tag=2772323005 To: Call-ID: 1735147470@192.168.1.91 CSeq: 1001 INVITE Contact: Proxy-Authorization: Digest username="550", realm="asterisk", nonce="0db53aaa", uri="sip:204@24.123.23.170:5060", response="ba9beb56aae61596b76b2f8cbdf18924", algorithm=MD5 max-forwards: 70 user-agent: UTSTARCOM F3000/Device ID-F3000_TEST Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, MESSAGE, REGISTER, REFER, PING, NOTIFY, INFO, PRACK Content-Type: application/sdp Content-Length: 298 v=0 o=550 123456 654322 IN IP4 192.168.1.91 s=none c=IN IP4 192.168.1.91 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 <-------------> [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: INVITE sip:204@24.123.23.170:5060 SIP/2.0 (41) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.91:5060;rport;branch=z9hG4bK4050918756 (65) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: "Display Name" ;tag=2772323005 (64) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (32) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 1735147470@192.168.1.91 (32) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: CSeq: 1001 INVITE (17) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: (36) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: Proxy-Authorization: Digest username="550", realm="asterisk", nonce="0db53aaa", uri="sip:204@24.123.23.170:5060", response="ba9beb56aae61596b76b2f8cbdf18924", algorithm=MD5 (172) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: max-forwards: 70 (16) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: user-agent: UTSTARCOM F3000/Device ID-F3000_TEST (48) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, MESSAGE, REGISTER, REFER, PING, NOTIFY, INFO, PRACK (93) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Content-Type: application/sdp (29) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: Content-Length: 298 (21) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 13: (0) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: v=0 (3) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: o=550 123456 654322 IN IP4 192.168.1.91 (39) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: s=none (6) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: c=IN IP4 192.168.1.91 (21) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: t=0 0 (5) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: m=audio 10010 RTP/AVP 0 8 18 2 101 (34) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=ptime:20 (10) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:18 G729A/8000 (22) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=fmtp:18 annexb=no (19) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:2 G726-32/8000 (23) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=fmtp:101 0-15 (15) [Feb 22 17:24:09] --- (13 headers 14 lines) --- [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Feb 22 17:24:09] Sending to 192.168.1.91 : 5060 (NAT) [Feb 22 17:24:09] Using INVITE request as basis request - 1735147470@192.168.1.91 [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:2559 do_setnat: Setting NAT on RTP to On [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:2564 do_setnat: Setting NAT on VRTP to On [Feb 22 17:24:09] Found user '550' [Feb 22 17:24:09] Found RTP audio format 0 [Feb 22 17:24:09] Found RTP audio format 8 [Feb 22 17:24:09] Found RTP audio format 18 [Feb 22 17:24:09] Found RTP audio format 2 [Feb 22 17:24:09] Found RTP audio format 101 [Feb 22 17:24:09] Peer audio RTP is at port 192.168.1.91:10010 [Feb 22 17:24:09] Found description format PCMU for ID 0 [Feb 22 17:24:09] Found description format PCMA for ID 8 [Feb 22 17:24:09] Found description format G729A for ID 18 [Feb 22 17:24:09] Got unsupported a:fmtp in SDP offer [Feb 22 17:24:09] Found description format G726-32 for ID 2 [Feb 22 17:24:09] Found description format telephone-event for ID 101 [Feb 22 17:24:09] Got unsupported a:fmtp in SDP offer [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:5068 process_sdp: T38 state changed to 0 on channel [Feb 22 17:24:09] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x90c (ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) [Feb 22 17:24:09] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Feb 22 17:24:09] Peer audio RTP is at port 192.168.1.91:10010 [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:5145 process_sdp: We're settling with these formats: 0xc (ulaw|alaw) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:13237 handle_request_invite: Checking SIP call limits for device 550 [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:2978 update_call_counter: Updating call counter for incoming call [Feb 22 17:24:09] Looking for 204 in smvoice-sip (domain 24.123.23.170) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:3766 sip_new: *** Our native formats are 0x4 (ulaw) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:3767 sip_new: *** Joint capabilities are 0xc (ulaw|alaw) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:3768 sip_new: *** Our capabilities are 0xe (gsm|ulaw|alaw) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:3769 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:3792 sip_new: This channel will not be able to handle video. [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:7891 build_route: build_route: Contact hop: [Feb 22 17:24:09] list_route: hop: [Feb 22 17:24:09] DEBUG[14767]: chan_sip.c:13310 handle_request_invite: SIP/550-09e07bc8: New call is still down.... Trying... [Feb 22 17:24:09] <--- Transmitting (NAT) to 192.168.1.91:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.91:5060;branch=z9hG4bK4050918756;received=192.168.1.91;rport=5060 From: "Display Name" ;tag=2772323005 To: Call-ID: 1735147470@192.168.1.91 CSeq: 1001 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:24:09] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/550-09e07bc8 [Feb 22 17:24:09] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 550 [Feb 22 17:24:09] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 550 [Feb 22 17:24:09] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/550 - state 1 (Not in use) [Feb 22 17:24:09] DEBUG[14829]: app_queue.c:546 changethread: Device 'SIP/550' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Feb 22 17:24:09] DEBUG[14828]: pbx.c:1767 pbx_extension_helper: Launching 'NoOp' [Feb 22 17:24:09] -- Executing [204@smvoice-sip:1] NoOp("SIP/550-09e07bc8", "2xx") in new stack [Feb 22 17:24:09] DEBUG[14828]: pbx.c:1767 pbx_extension_helper: Launching 'Set' [Feb 22 17:24:09] -- Executing [204@smvoice-sip:2] Set("SIP/550-09e07bc8", "SMVOICE_CONTEXT_EXTEN=204") in new stack [Feb 22 17:24:09] DEBUG[14828]: pbx.c:1767 pbx_extension_helper: Launching 'AGI' [Feb 22 17:24:09] -- Executing [204@smvoice-sip:3] AGI("SIP/550-09e07bc8", "smvoice|-company|demonstration|-digium_asterisk|-asterisk_callat_forwarding|204") in new stack [Feb 22 17:24:09] -- Launched AGI Script /var/lib/asterisk/agi-bin/smvoice [Feb 22 17:24:09] -- AGI Script smvoice completed, returning 0 [Feb 22 17:24:09] DEBUG[14828]: pbx.c:1688 pbx_substitute_variables_helper_full: Expression result is '0' [Feb 22 17:24:09] DEBUG[14828]: pbx.c:1767 pbx_extension_helper: Launching 'GotoIf' [Feb 22 17:24:09] -- Executing [204@smvoice-sip:4] GotoIf("SIP/550-09e07bc8", "0?INVALID|1") in new stack [Feb 22 17:24:09] DEBUG[14828]: pbx.c:5932 pbx_builtin_gotoif: Not taking any branch [Feb 22 17:24:09] DEBUG[14828]: pbx.c:1688 pbx_substitute_variables_helper_full: Expression result is '0' [Feb 22 17:24:09] DEBUG[14828]: pbx.c:1767 pbx_extension_helper: Launching 'GotoIf' [Feb 22 17:24:09] -- Executing [204@smvoice-sip:5] GotoIf("SIP/550-09e07bc8", "0?_2XX-NOANSWER|1") in new stack [Feb 22 17:24:09] DEBUG[14828]: pbx.c:5932 pbx_builtin_gotoif: Not taking any branch [Feb 22 17:24:09] DEBUG[14828]: pbx.c:1767 pbx_extension_helper: Launching 'Dial' [Feb 22 17:24:09] -- Executing [204@smvoice-sip:6] Dial("SIP/550-09e07bc8", "SIP/528|20|") in new stack [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:15106 sip_request_call: Asked to create a SIP channel with formats: 0x4 (ulaw) [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:2559 do_setnat: Setting NAT on RTP to Off [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:3766 sip_new: *** Our native formats are 0x4 (ulaw) [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:3767 sip_new: *** Joint capabilities are 0x0 (nothing) [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:3768 sip_new: *** Our capabilities are 0xe (gsm|ulaw|alaw) [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:3769 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:3771 sip_new: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:3792 sip_new: This channel will not be able to handle video. [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:16701 sip_set_rtp_peer: Early remote bridge setting SIP '1ca0c9900034b45264c7ee1a67dc712c@24.123.23.170' - Sending media to 192.168.1.91 [Feb 22 17:24:09] DEBUG[14828]: rtp.c:1570 ast_rtp_make_compatible: Seeded SDP of 'SIP/528-09e0cbb8' with that of 'SIP/550-09e07bc8' [Feb 22 17:24:09] DEBUG[14828]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-smvoice-sip-204-6. [Feb 22 17:24:09] DEBUG[14828]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-smvoice-sip-204-5. [Feb 22 17:24:09] DEBUG[14828]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-smvoice-sip-204-4. [Feb 22 17:24:09] DEBUG[14828]: channel.c:3185 ast_channel_inherit_variables: Not copying variable AGISTATUS. [Feb 22 17:24:09] DEBUG[14828]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SMVOICE_WORK_EXTENSION. [Feb 22 17:24:09] DEBUG[14828]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SMVOICE_EXTENSION. [Feb 22 17:24:09] DEBUG[14828]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SMVOICE_PIN. [Feb 22 17:24:09] DEBUG[14828]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SMVOICE_CALLAT. [Feb 22 17:24:09] DEBUG[14828]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-smvoice-sip-204-3. [Feb 22 17:24:09] DEBUG[14828]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SMVOICE_CONTEXT_EXTEN. [Feb 22 17:24:09] DEBUG[14828]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-smvoice-sip-204-2. [Feb 22 17:24:09] DEBUG[14828]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-smvoice-sip-204-1. [Feb 22 17:24:09] DEBUG[14828]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Feb 22 17:24:09] DEBUG[14828]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT. [Feb 22 17:24:09] DEBUG[14828]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Feb 22 17:24:09] DEBUG[14828]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPURI. [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:2812 sip_call: Outgoing Call for 528 [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:2978 update_call_counter: Updating call counter for outgoing call [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:2826 sip_call: Our T38 capability (0), joint T38 capability (0) [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:6122 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: False [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:6123 add_sdp: ** Our prefcodec: 0x4 (ulaw) [Feb 22 17:24:09] Audio is at 24.123.23.170 port 19576 [Feb 22 17:24:09] Adding codec 0x4 (ulaw) to SDP [Feb 22 17:24:09] Adding codec 0x8 (alaw) to SDP [Feb 22 17:24:09] Adding codec 0x2 (gsm) to SDP [Feb 22 17:24:09] Adding non-codec 0x1 (telephone-event) to SDP [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:6254 add_sdp: -- Done with adding codecs to SDP [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:6299 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 0: INVITE sip:528@192.168.1.93:5060 SIP/2.0 (40) [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5827fbbd (58) [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 2: From: "550 550" ;tag=as239b687a (54) [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 3: To: (31) [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 4: Contact: (32) [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 5: Call-ID: 1ca0c9900034b45264c7ee1a67dc712c@24.123.23.170 (55) [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 6: CSeq: 102 INVITE (16) [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 9: Date: Thu, 22 Feb 2007 22:24:09 GMT (35) [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 11: Supported: replaces (19) [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 12: Content-Type: application/sdp (29) [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 13: Content-Length: 287 (19) [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 14: (0) [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:4559 parse_request: Line: v=0 (3) [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:4559 parse_request: Line: o=root 14736 14736 IN IP4 192.168.1.91 (38) [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:4559 parse_request: Line: s=session (9) [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:4559 parse_request: Line: c=IN IP4 192.168.1.91 (21) [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:4559 parse_request: Line: t=0 0 (5) [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:4559 parse_request: Line: m=audio 10010 RTP/AVP 0 8 3 101 (31) [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:4559 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:4559 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:4559 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:4559 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:4559 parse_request: Line: a=fmtp:101 0-16 (15) [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:4559 parse_request: Line: a=silenceSupp:off - - - - (25) [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:4559 parse_request: Line: a=ptime:20 (10) [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:4559 parse_request: Line: a=sendrecv (10) [Feb 22 17:24:09] Reliably Transmitting (no NAT) to 192.168.1.93:5060: INVITE sip:528@192.168.1.93:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5827fbbd From: "550 550" ;tag=as239b687a To: Contact: Call-ID: 1ca0c9900034b45264c7ee1a67dc712c@24.123.23.170 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 22 Feb 2007 22:24:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 287 v=0 o=root 14736 14736 IN IP4 192.168.1.91 s=session c=IN IP4 192.168.1.91 t=0 0 m=audio 10010 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Feb 22 17:24:09] DEBUG[14828]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #182 [Feb 22 17:24:09] -- Called 528 [Feb 22 17:24:10] <--- SIP read from 192.168.1.93:5060 ---> SIP/2.0 100 Trying To: ;tag=177014cec8654efa2f4863c390adf2a0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5827fbbd From: "550 550" ;tag=as239b687a Call-ID: 1ca0c9900034b45264c7ee1a67dc712c@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 <-------------> [Feb 22 17:24:10] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 100 Trying (18) [Feb 22 17:24:10] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: To: ;tag=177014cec8654efa2f4863c390adf2a0 (68) [Feb 22 17:24:10] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5827fbbd (58) [Feb 22 17:24:10] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: From: "550 550" ;tag=as239b687a (54) [Feb 22 17:24:10] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 1ca0c9900034b45264c7ee1a67dc712c@24.123.23.170 (55) [Feb 22 17:24:10] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: CSeq: 102 INVITE (16) [Feb 22 17:24:10] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Content-Length: 0 (17) [Feb 22 17:24:10] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:24:10] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:24:10] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: (0) [Feb 22 17:24:10] --- (9 headers 0 lines) --- [Feb 22 17:24:10] DEBUG[14767]: chan_sip.c:2104 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #182 - INVITE (got response) [Feb 22 17:24:10] DEBUG[14767]: chan_sip.c:2113 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '1ca0c9900034b45264c7ee1a67dc712c@24.123.23.170' Request 102: Found [Feb 22 17:24:10] DEBUG[14767]: chan_sip.c:11531 handle_response_invite: SIP response 100 to standard invite [Feb 22 17:24:10] <--- SIP read from 192.168.1.93:5060 ---> SIP/2.0 180 Ringing To: ;tag=177014cec8654efa2f4863c390adf2a0 Contact: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5827fbbd From: "550 550" ;tag=as239b687a Call-ID: 1ca0c9900034b45264c7ee1a67dc712c@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 <-------------> [Feb 22 17:24:10] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 180 Ringing (19) [Feb 22 17:24:10] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: To: ;tag=177014cec8654efa2f4863c390adf2a0 (68) [Feb 22 17:24:10] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: Contact: (36) [Feb 22 17:24:10] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5827fbbd (58) [Feb 22 17:24:10] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: "550 550" ;tag=as239b687a (54) [Feb 22 17:24:10] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Call-ID: 1ca0c9900034b45264c7ee1a67dc712c@24.123.23.170 (55) [Feb 22 17:24:10] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: CSeq: 102 INVITE (16) [Feb 22 17:24:10] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: Content-Length: 0 (17) [Feb 22 17:24:10] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:24:10] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:24:10] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: (0) [Feb 22 17:24:10] --- (10 headers 0 lines) --- [Feb 22 17:24:10] DEBUG[14767]: chan_sip.c:2113 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '1ca0c9900034b45264c7ee1a67dc712c@24.123.23.170' Request 102: Found [Feb 22 17:24:10] DEBUG[14767]: chan_sip.c:11531 handle_response_invite: SIP response 180 to standard invite [Feb 22 17:24:10] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/528-09e0cbb8 [Feb 22 17:24:10] -- SIP/528-09e0cbb8 is ringing [Feb 22 17:24:10] DEBUG[14828]: rtp.c:1505 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/550-09e07bc8' with that of 'SIP/528-09e0cbb8' [Feb 22 17:24:10] <--- Transmitting (NAT) to 192.168.1.91:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.91:5060;branch=z9hG4bK4050918756;received=192.168.1.91;rport=5060 From: "Display Name" ;tag=2772323005 To: ;tag=as2b86d5e2 Call-ID: 1735147470@192.168.1.91 CSeq: 1001 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:24:10] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 528 [Feb 22 17:24:10] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 528 [Feb 22 17:24:10] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/528 - state 1 (Not in use) [Feb 22 17:24:10] DEBUG[14831]: app_queue.c:546 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. [Feb 22 17:24:12] DEBUG[14767]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '55ae66469f11@24.123.23.170' [Feb 22 17:24:12] DEBUG[14767]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 55ae66469f11@24.123.23.170 [Feb 22 17:24:12] Really destroying SIP dialog '55ae66469f11@24.123.23.170' Method: REGISTER [Feb 22 17:24:13] <--- SIP read from 192.168.1.93:5060 ---> INVITE sip:526@24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKr770111ef2477c443c967f425385e31f7 Call-ID: 524091ef199964d13508e0184586dc1c@24.123.23.170 CSeq: 440468 INVITE From: 528 ;tag=81c5c0cdcd26df8f2948fc3e1971f0eb To: ;tag=as449238ce Contact: 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: 241 v=0 o=- 14736 70057 IN IP4 192.168.1.93 s=session c=IN IP4 0.0.0.0 t=0 0 m=audio 30030 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 - - - - a=ptime:20 a=sendonly <-------------> [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: INVITE sip:526@24.123.23.170 SIP/2.0 (36) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKr770111ef2477c443c967f425385e31f7 (82) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: Call-ID: 524091ef199964d13508e0184586dc1c@24.123.23.170 (55) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: CSeq: 440468 INVITE (19) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: 528 ;tag=81c5c0cdcd26df8f2948fc3e1971f0eb (74) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: To: ;tag=as449238ce (42) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: (36) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: Content-Type: application/sdp (29) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Max-Forwards: 70 (16) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: Content-Length: 241 (19) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 13: (0) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: v=0 (3) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: o=- 14736 70057 IN IP4 192.168.1.93 (35) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: s=session (9) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: c=IN IP4 0.0.0.0 (16) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: t=0 0 (5) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: m=audio 30030 RTP/AVP 0 101 (27) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=fmtp:101 0-15,16,17,18 (24) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=silenceSupp:off - - - - (25) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=ptime:20 (10) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=sendonly (10) [Feb 22 17:24:13] --- (13 headers 12 lines) --- [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:1662 parse_sip_options: Begin: parsing SIP "Supported: sip-cc, sip-cc-01, replaces, timer" [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:1670 parse_sip_options: Found SIP option: -sip-cc- [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:1684 parse_sip_options: Found no match for SIP option: sip-cc (Please file bug report!) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:1670 parse_sip_options: Found SIP option: -sip-cc-01- [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:1684 parse_sip_options: Found no match for SIP option: sip-cc-01 (Please file bug report!) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:1670 parse_sip_options: Found SIP option: -replaces- [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:1676 parse_sip_options: Matched SIP option: replaces [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:1670 parse_sip_options: Found SIP option: -timer- [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:1676 parse_sip_options: Matched SIP option: timer [Feb 22 17:24:13] Sending to 192.168.1.93 : 5060 (no NAT) [Feb 22 17:24:13] Found RTP audio format 0 [Feb 22 17:24:13] Found RTP audio format 101 [Feb 22 17:24:13] Peer audio RTP is at port 0.0.0.0:30030 [Feb 22 17:24:13] Found description format PCMU for ID 0 [Feb 22 17:24:13] Found description format telephone-event for ID 101 [Feb 22 17:24:13] Got unsupported a:fmtp in SDP offer [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:5068 process_sdp: T38 state changed to 0 on channel SIP/528-09e02620 [Feb 22 17:24:13] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Feb 22 17:24:13] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Feb 22 17:24:13] Peer audio RTP is at port 0.0.0.0:30030 [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:5145 process_sdp: We're settling with these formats: 0x4 (ulaw) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:5152 process_sdp: We have an owner, now see if we need to change this call [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:13288 handle_request_invite: Got a SIP re-invite for call 524091ef199964d13508e0184586dc1c@24.123.23.170 [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:13383 handle_request_invite: SIP/528-09e02620: This call is UP.... [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:6354 transmit_response_with_sdp: Setting framing from config on incoming call [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:6122 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: True [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:6123 add_sdp: ** Our prefcodec: 0x4 (ulaw) [Feb 22 17:24:13] Audio is at 24.123.23.170 port 16212 [Feb 22 17:24:13] Adding codec 0x4 (ulaw) to SDP [Feb 22 17:24:13] Adding non-codec 0x1 (telephone-event) to SDP [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:6254 add_sdp: -- Done with adding codecs to SDP [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:6299 add_sdp: Done building SDP. Settling with this capability: 0x4 (ulaw) [Feb 22 17:24:13] <--- Reliably 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=z9hG4bKr770111ef2477c443c967f425385e31f7;received=192.168.1.93 From: 528 ;tag=81c5c0cdcd26df8f2948fc3e1971f0eb To: ;tag=as449238ce Call-ID: 524091ef199964d13508e0184586dc1c@24.123.23.170 CSeq: 440468 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 240 v=0 o=root 14736 14737 IN IP4 192.168.1.66 s=session c=IN IP4 192.168.1.66 t=0 0 m=audio 30006 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #184 [Feb 22 17:24:13] DEBUG[14821]: channel.c:2731 set_format: Set channel SIP/526-09df30e0 to write format slin [Feb 22 17:24:13] -- Started music on hold, class 'default', on channel 'SIP/526-09df30e0' [Feb 22 17:24:13] DEBUG[14821]: channel.c:1908 ast_settimeout: Scheduling timer at 160 sample intervals [Feb 22 17:24:13] DEBUG[14821]: rtp.c:2783 bridge_native_loop: Oooh, 'SIP/528-09e02620' changed end address to 0.0.0.0:0 (format 4) [Feb 22 17:24:13] DEBUG[14821]: rtp.c:2785 bridge_native_loop: Oooh, 'SIP/528-09e02620' changed end vaddress to 0.0.0.0:0 (format 4) [Feb 22 17:24:13] DEBUG[14821]: rtp.c:2787 bridge_native_loop: Oooh, 'SIP/528-09e02620' was 192.168.1.93:30030/(format 4) [Feb 22 17:24:13] DEBUG[14821]: rtp.c:2789 bridge_native_loop: Oooh, 'SIP/528-09e02620' was 0.0.0.0:0/(format 4) [Feb 22 17:24:13] DEBUG[14821]: chan_sip.c:16704 sip_set_rtp_peer: Sending reinvite on SIP '903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170' - It's audio soon redirected to IP 24.123.23.170 [Feb 22 17:24:13] DEBUG[14821]: chan_sip.c:5576 reqprep: Strict routing enforced for session 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 [Feb 22 17:24:13] set_destination: Parsing for address/port to send to [Feb 22 17:24:13] set_destination: set destination to 192.168.1.66, port 5060 [Feb 22 17:24:13] DEBUG[14821]: chan_sip.c:6122 add_sdp: ** Our capability: 0xc (ulaw|alaw) Video flag: True [Feb 22 17:24:13] DEBUG[14821]: chan_sip.c:6123 add_sdp: ** Our prefcodec: 0x0 (nothing) [Feb 22 17:24:13] Audio is at 24.123.23.170 port 15904 [Feb 22 17:24:13] Adding codec 0x4 (ulaw) to SDP [Feb 22 17:24:13] Adding codec 0x8 (alaw) to SDP [Feb 22 17:24:13] Adding non-codec 0x1 (telephone-event) to SDP [Feb 22 17:24:13] DEBUG[14821]: chan_sip.c:6254 add_sdp: -- Done with adding codecs to SDP [Feb 22 17:24:13] DEBUG[14821]: chan_sip.c:6299 add_sdp: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Feb 22 17:24:13] DEBUG[14821]: chan_sip.c:1603 initialize_initreq: Initializing already initialized SIP dialog 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 (presumably reinvite) [Feb 22 17:24:13] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 0: INVITE sip:526@192.168.1.66:5060 SIP/2.0 (40) [Feb 22 17:24:13] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5f8b21f1 (58) [Feb 22 17:24:13] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 2: From: ;tag=as056fdf38 (44) [Feb 22 17:24:13] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 3: To: 526 ;tag=48106fce45713fff4b942431e90afa0e (68) [Feb 22 17:24:13] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 4: Contact: (32) [Feb 22 17:24:13] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 5: Call-ID: 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 (68) [Feb 22 17:24:13] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 6: CSeq: 102 INVITE (16) [Feb 22 17:24:13] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Feb 22 17:24:13] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 22 17:24:13] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Feb 22 17:24:13] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 10: Supported: replaces (19) [Feb 22 17:24:13] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Feb 22 17:24:13] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 12: Content-Type: application/sdp (29) [Feb 22 17:24:13] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 13: Content-Length: 266 (19) [Feb 22 17:24:13] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 14: (0) [Feb 22 17:24:13] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: v=0 (3) [Feb 22 17:24:13] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: o=root 14736 14737 IN IP4 24.123.23.170 (39) [Feb 22 17:24:13] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: s=session (9) [Feb 22 17:24:13] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: c=IN IP4 24.123.23.170 (22) [Feb 22 17:24:13] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: t=0 0 (5) [Feb 22 17:24:13] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: m=audio 15904 RTP/AVP 0 8 101 (29) [Feb 22 17:24:13] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Feb 22 17:24:13] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Feb 22 17:24:13] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Feb 22 17:24:13] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: a=fmtp:101 0-16 (15) [Feb 22 17:24:13] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: a=silenceSupp:off - - - - (25) [Feb 22 17:24:13] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: a=ptime:20 (10) [Feb 22 17:24:13] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: a=sendrecv (10) [Feb 22 17:24:13] Reliably Transmitting (no NAT) to 192.168.1.66:5060: INVITE sip:526@192.168.1.66:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5f8b21f1 From: ;tag=as056fdf38 To: 526 ;tag=48106fce45713fff4b942431e90afa0e Contact: Call-ID: 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 266 v=0 o=root 14736 14737 IN IP4 24.123.23.170 s=session c=IN IP4 24.123.23.170 t=0 0 m=audio 15904 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 - - - - a=ptime:20 a=sendrecv --- [Feb 22 17:24:13] DEBUG[14821]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #185 [Feb 22 17:24:13] DEBUG[14821]: channel.c:1689 generator_force: Auto-deactivating generator [Feb 22 17:24:13] DEBUG[14821]: channel.c:2731 set_format: Set channel SIP/526-09df30e0 to write format ulaw [Feb 22 17:24:13] -- Stopped music on hold on SIP/526-09df30e0 [Feb 22 17:24:13] DEBUG[14821]: channel.c:1908 ast_settimeout: Scheduling timer at 0 sample intervals [Feb 22 17:24:13] <--- SIP read from 192.168.1.93:5060 ---> ACK sip:526@24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKv8f3e33daf717f7914e07df2b6158f0ae CSeq: 440468 ACK To: ;tag=as449238ce Call-ID: 524091ef199964d13508e0184586dc1c@24.123.23.170 From: 528 ;tag=81c5c0cdcd26df8f2948fc3e1971f0eb User-Agent: Uniden SIP Phone p2 Ver BS4.77 <-------------> [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: ACK sip:526@24.123.23.170 SIP/2.0 (33) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKv8f3e33daf717f7914e07df2b6158f0ae (82) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: CSeq: 440468 ACK (16) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: ;tag=as449238ce (42) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 524091ef199964d13508e0184586dc1c@24.123.23.170 (55) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: From: 528 ;tag=81c5c0cdcd26df8f2948fc3e1971f0eb (74) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: (0) [Feb 22 17:24:13] --- (7 headers 0 lines) --- [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received ACK (6) - Command in SIP ACK [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:2061 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #184 [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '524091ef199964d13508e0184586dc1c@24.123.23.170' of Response 440468: Match Not Found [Feb 22 17:24:13] <--- SIP read from 192.168.1.66:5060 ---> SIP/2.0 200 OK To: 526 ;tag=48106fce45713fff4b942431e90afa0e Contact: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5f8b21f1 From: ;tag=as056fdf38 Call-ID: 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 CSeq: 102 INVITE Content-Type: application/sdp Content-Length: 247 Supported: sip-cc, sip-cc-01, replaces, timer Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK v=0 o=- 14736 163216 IN IP4 192.168.1.66 s=session c=IN IP4 192.168.1.66 t=0 0 m=audio 30006 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 - - - - a=ptime:20 a=sendrecv <-------------> [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: To: 526 ;tag=48106fce45713fff4b942431e90afa0e (68) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: Contact: (36) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5f8b21f1 (58) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: ;tag=as056fdf38 (44) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Call-ID: 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 (68) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: CSeq: 102 INVITE (16) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: Content-Type: application/sdp (29) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Content-Length: 247 (19) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: (0) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: v=0 (3) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: o=- 14736 163216 IN IP4 192.168.1.66 (36) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: s=session (9) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: c=IN IP4 192.168.1.66 (21) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: t=0 0 (5) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: m=audio 30006 RTP/AVP 0 101 (27) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=fmtp:101 0-15,16,17,18 (24) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=silenceSupp:off - - - - (25) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=ptime:20 (10) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=sendrecv (10) [Feb 22 17:24:13] --- (11 headers 12 lines) --- [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:2053 __sip_ack: Acked pending invite 102 [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:2061 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #185 [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170' of Request 102: Match Not Found [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:11529 handle_response_invite: SIP response 200 to RE-invite on outgoing call 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 [Feb 22 17:24:13] Found RTP audio format 0 [Feb 22 17:24:13] Found RTP audio format 101 [Feb 22 17:24:13] Peer audio RTP is at port 192.168.1.66:30006 [Feb 22 17:24:13] Found description format PCMU for ID 0 [Feb 22 17:24:13] Found description format telephone-event for ID 101 [Feb 22 17:24:13] Got unsupported a:fmtp in SDP offer [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:5068 process_sdp: T38 state changed to 0 on channel SIP/526-09df30e0 [Feb 22 17:24:13] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Feb 22 17:24:13] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Feb 22 17:24:13] Peer audio RTP is at port 192.168.1.66:30006 [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:5145 process_sdp: We're settling with these formats: 0x4 (ulaw) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:5152 process_sdp: We have an owner, now see if we need to change this call [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:2978 update_call_counter: Updating call counter for outgoing call [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:7891 build_route: build_route: Contact hop: [Feb 22 17:24:13] list_route: hop: [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:11662 handle_response_invite: Strange... The other side of the bridge does not have a udptl struct [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:11667 handle_response_invite: T38 state changed to 0 on channel SIP [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:11670 handle_response_invite: T38 state changed to 0 on channel SIP/526-09df30e0 [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:5576 reqprep: Strict routing enforced for session 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 [Feb 22 17:24:13] set_destination: Parsing for address/port to send to [Feb 22 17:24:13] set_destination: set destination to 192.168.1.66, port 5060 [Feb 22 17:24:13] Transmitting (no NAT) to 192.168.1.66:5060: ACK sip:526@192.168.1.66:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK4bc4b355 From: ;tag=as056fdf38 To: 526 ;tag=48106fce45713fff4b942431e90afa0e Contact: Call-ID: 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Feb 22 17:24:13] <--- SIP read from 192.168.1.93:5060 ---> SIP/2.0 200 OK To: ;tag=177014cec8654efa2f4863c390adf2a0 Contact: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5827fbbd From: "550 550" ;tag=as239b687a Call-ID: 1ca0c9900034b45264c7ee1a67dc712c@24.123.23.170 CSeq: 102 INVITE Content-Type: application/sdp Content-Length: 247 Supported: sip-cc, sip-cc-01, replaces, timer Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK v=0 o=- 14736 345898 IN IP4 192.168.1.93 s=session c=IN IP4 192.168.1.93 t=0 0 m=audio 30032 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 - - - - a=ptime:20 a=sendrecv <-------------> [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: To: ;tag=177014cec8654efa2f4863c390adf2a0 (68) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: Contact: (36) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5827fbbd (58) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: "550 550" ;tag=as239b687a (54) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Call-ID: 1ca0c9900034b45264c7ee1a67dc712c@24.123.23.170 (55) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: CSeq: 102 INVITE (16) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: Content-Type: application/sdp (29) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Content-Length: 247 (19) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: (0) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: v=0 (3) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: o=- 14736 345898 IN IP4 192.168.1.93 (36) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: s=session (9) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: c=IN IP4 192.168.1.93 (21) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: t=0 0 (5) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: m=audio 30032 RTP/AVP 0 101 (27) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=fmtp:101 0-15,16,17,18 (24) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=silenceSupp:off - - - - (25) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=ptime:20 (10) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=sendrecv (10) [Feb 22 17:24:13] --- (11 headers 12 lines) --- [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:2053 __sip_ack: Acked pending invite 102 [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '1ca0c9900034b45264c7ee1a67dc712c@24.123.23.170' of Request 102: Match Not Found [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:11531 handle_response_invite: SIP response 200 to standard invite [Feb 22 17:24:13] Found RTP audio format 0 [Feb 22 17:24:13] Found RTP audio format 101 [Feb 22 17:24:13] Peer audio RTP is at port 192.168.1.93:30032 [Feb 22 17:24:13] Found description format PCMU for ID 0 [Feb 22 17:24:13] Found description format telephone-event for ID 101 [Feb 22 17:24:13] Got unsupported a:fmtp in SDP offer [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:5068 process_sdp: T38 state changed to 0 on channel SIP/528-09e0cbb8 [Feb 22 17:24:13] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Feb 22 17:24:13] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Feb 22 17:24:13] Peer audio RTP is at port 192.168.1.93:30032 [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:5145 process_sdp: We're settling with these formats: 0x4 (ulaw) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:5152 process_sdp: We have an owner, now see if we need to change this call [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:2978 update_call_counter: Updating call counter for outgoing call [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:7891 build_route: build_route: Contact hop: [Feb 22 17:24:13] list_route: hop: [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:5576 reqprep: Strict routing enforced for session 1ca0c9900034b45264c7ee1a67dc712c@24.123.23.170 [Feb 22 17:24:13] set_destination: Parsing for address/port to send to [Feb 22 17:24:13] set_destination: set destination to 192.168.1.93, port 5060 [Feb 22 17:24:13] Transmitting (no NAT) to 192.168.1.93:5060: ACK sip:528@192.168.1.93:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK7bd8c7d4 From: "550 550" ;tag=as239b687a To: ;tag=177014cec8654efa2f4863c390adf2a0 Contact: Call-ID: 1ca0c9900034b45264c7ee1a67dc712c@24.123.23.170 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Feb 22 17:24:13] DEBUG[14828]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/528-09e0cbb8 [Feb 22 17:24:13] -- SIP/528-09e0cbb8 answered SIP/550-09e07bc8 [Feb 22 17:24:13] DEBUG[14828]: chan_sip.c:16701 sip_set_rtp_peer: Early remote bridge setting SIP '1735147470@192.168.1.91' - Sending media to 192.168.1.93 [Feb 22 17:24:13] DEBUG[14828]: rtp.c:1505 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/550-09e07bc8' with that of 'SIP/528-09e0cbb8' [Feb 22 17:24:13] DEBUG[14828]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/550-09e07bc8 [Feb 22 17:24:13] DEBUG[14828]: chan_sip.c:3428 sip_answer: SIP answering channel: SIP/550-09e07bc8 [Feb 22 17:24:13] DEBUG[14828]: chan_sip.c:6354 transmit_response_with_sdp: Setting framing from config on incoming call [Feb 22 17:24:13] DEBUG[14828]: chan_sip.c:6122 add_sdp: ** Our capability: 0xc (ulaw|alaw) Video flag: True [Feb 22 17:24:13] DEBUG[14828]: chan_sip.c:6123 add_sdp: ** Our prefcodec: 0x0 (nothing) [Feb 22 17:24:13] Audio is at 24.123.23.170 port 15940 [Feb 22 17:24:13] Adding codec 0x4 (ulaw) to SDP [Feb 22 17:24:13] Adding codec 0x8 (alaw) to SDP [Feb 22 17:24:13] Adding non-codec 0x1 (telephone-event) to SDP [Feb 22 17:24:13] DEBUG[14828]: chan_sip.c:6254 add_sdp: -- Done with adding codecs to SDP [Feb 22 17:24:13] DEBUG[14828]: chan_sip.c:6299 add_sdp: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Feb 22 17:24:13] <--- Reliably Transmitting (NAT) to 192.168.1.91:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.91:5060;branch=z9hG4bK4050918756;received=192.168.1.91;rport=5060 From: "Display Name" ;tag=2772323005 To: ;tag=as2b86d5e2 Call-ID: 1735147470@192.168.1.91 CSeq: 1001 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 264 v=0 o=root 14736 14736 IN IP4 192.168.1.93 s=session c=IN IP4 192.168.1.93 t=0 0 m=audio 30032 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 - - - - a=ptime:20 a=sendrecv <------------> [Feb 22 17:24:13] DEBUG[14828]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #187 [Feb 22 17:24:13] -- Native bridging SIP/550-09e07bc8 and SIP/528-09e0cbb8 [Feb 22 17:24:13] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 528 [Feb 22 17:24:13] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 528 [Feb 22 17:24:13] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/528 - state 1 (Not in use) [Feb 22 17:24:13] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 550 [Feb 22 17:24:13] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 550 [Feb 22 17:24:13] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/550 - state 1 (Not in use) [Feb 22 17:24:13] DEBUG[14832]: app_queue.c:546 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. [Feb 22 17:24:13] DEBUG[14833]: app_queue.c:546 changethread: Device 'SIP/550' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Feb 22 17:24:13] <--- SIP read from 192.168.1.91:5060 ---> ACK sip:204@24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.91:5060;rport;branch=z9hG4bK2470845385 From: "Display Name" ;tag=2772323005 To: ;tag=as2b86d5e2 Call-ID: 1735147470@192.168.1.91 CSeq: 1001 ACK max-forwards: 70 user-agent: UTSTARCOM F3000/Device ID-F3000_TEST Content-Length: 0 <-------------> [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: ACK sip:204@24.123.23.170 SIP/2.0 (33) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.91:5060;rport;branch=z9hG4bK2470845385 (65) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: "Display Name" ;tag=2772323005 (64) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: ;tag=as2b86d5e2 (47) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 1735147470@192.168.1.91 (32) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: CSeq: 1001 ACK (14) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: max-forwards: 70 (16) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: user-agent: UTSTARCOM F3000/Device ID-F3000_TEST (48) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Content-Length: 0 (17) [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: (0) [Feb 22 17:24:13] --- (9 headers 0 lines) --- [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received ACK (6) - Command in SIP ACK [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:2061 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #187 [Feb 22 17:24:13] DEBUG[14767]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '1735147470@192.168.1.91' of Response 1001: Match Not Found [Feb 22 17:24:16] <--- SIP read from 192.168.1.86:5062 ---> <-------------> [Feb 22 17:24:16] --- (0 headers 0 lines) Nat keepalive --- [Feb 22 17:24:16] <--- 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=z9hG4bK3d808613 From: To: Call-ID: 000ff78d-ebb20002-237ee55f-6d451a9c@192.168.1.165 Date: Thu, 22 Feb 2007 22:25:50 GMT CSeq: 49716 REGISTER User-Agent: CSCO/7 Contact: Content-Length: 0 Expires: 60 <-------------> [Feb 22 17:24:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:24:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK3d808613 (58) [Feb 22 17:24:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: (40) [Feb 22 17:24:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (38) [Feb 22 17:24:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 000ff78d-ebb20002-237ee55f-6d451a9c@192.168.1.165 (58) [Feb 22 17:24:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Date: Thu, 22 Feb 2007 22:25:50 GMT (35) [Feb 22 17:24:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: CSeq: 49716 REGISTER (20) [Feb 22 17:24:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: CSCO/7 (18) [Feb 22 17:24:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Contact: (37) [Feb 22 17:24:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:24:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Expires: 60 (11) [Feb 22 17:24:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: (0) [Feb 22 17:24:16] --- (11 headers 0 lines) --- [Feb 22 17:24:16] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 000ff78d-ebb20002-237ee55f-6d451a9c@192.168.1.165 - REGISTER (No RTP) [Feb 22 17:24:16] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:24:16] Using latest REGISTER request as basis request [Feb 22 17:24:16] Sending to 192.168.1.165 : 5060 (no NAT) [Feb 22 17:24:16] <--- 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=z9hG4bK3d808613;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20002-237ee55f-6d451a9c@192.168.1.165 CSeq: 49716 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:24:16] <--- 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=z9hG4bK3d808613;received=192.168.1.165 From: To: ;tag=as434baad7 Call-ID: 000ff78d-ebb20002-237ee55f-6d451a9c@192.168.1.165 CSeq: 49716 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7cf0f418" Content-Length: 0 <------------> [Feb 22 17:24:16] Scheduling destruction of SIP dialog '000ff78d-ebb20002-237ee55f-6d451a9c@192.168.1.165' in 32000 ms (Method: REGISTER) [Feb 22 17:24:16] <--- 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=z9hG4bK72f43cbe From: To: Call-ID: 000ff78d-ebb20002-237ee55f-6d451a9c@192.168.1.165 Date: Thu, 22 Feb 2007 22:25:50 GMT CSeq: 49717 REGISTER User-Agent: CSCO/7 Contact: Authorization: Digest username="510",realm="asterisk",uri="sip:24.123.23.170",response="2ea3a049b42d558c96f6d3f1e21d8b7c",nonce="7cf0f418",algorithm=MD5 Content-Length: 0 Expires: 60 <-------------> [Feb 22 17:24:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:24:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK72f43cbe (58) [Feb 22 17:24:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: (40) [Feb 22 17:24:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (38) [Feb 22 17:24:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 000ff78d-ebb20002-237ee55f-6d451a9c@192.168.1.165 (58) [Feb 22 17:24:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Date: Thu, 22 Feb 2007 22:25:50 GMT (35) [Feb 22 17:24:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: CSeq: 49717 REGISTER (20) [Feb 22 17:24:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: CSCO/7 (18) [Feb 22 17:24:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Contact: (37) [Feb 22 17:24:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Authorization: Digest username="510",realm="asterisk",uri="sip:24.123.23.170",response="2ea3a049b42d558c96f6d3f1e21d8b7c",nonce="7cf0f418",algorithm=MD5 (152) [Feb 22 17:24:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Content-Length: 0 (17) [Feb 22 17:24:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Expires: 60 (11) [Feb 22 17:24:16] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: (0) [Feb 22 17:24:16] --- (12 headers 0 lines) --- [Feb 22 17:24:16] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:24:16] Using latest REGISTER request as basis request [Feb 22 17:24:16] Sending to 192.168.1.165 : 5060 (no NAT) [Feb 22 17:24:16] <--- 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=z9hG4bK72f43cbe;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20002-237ee55f-6d451a9c@192.168.1.165 CSeq: 49717 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:24:16] <--- 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=z9hG4bK72f43cbe;received=192.168.1.165 From: To: ;tag=as434baad7 Call-ID: 000ff78d-ebb20002-237ee55f-6d451a9c@192.168.1.165 CSeq: 49717 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Thu, 22 Feb 2007 22:24:16 GMT Content-Length: 0 <------------> [Feb 22 17:24:16] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/510 [Feb 22 17:24:16] Scheduling destruction of SIP dialog '000ff78d-ebb20002-237ee55f-6d451a9c@192.168.1.165' in 32000 ms (Method: REGISTER) [Feb 22 17:24:16] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 510 [Feb 22 17:24:16] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 510 [Feb 22 17:24:16] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/510 - state 1 (Not in use) [Feb 22 17:24:16] DEBUG[14834]: app_queue.c:546 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. [Feb 22 17:24:17] <--- 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=z9hG4bK7f8eb05f From: To: Call-ID: 000ff78d-ebb20004-75d3ddb3-71096c5c@192.168.1.165 Date: Thu, 22 Feb 2007 22:25:51 GMT CSeq: 49692 REGISTER User-Agent: CSCO/7 Contact: Content-Length: 0 Expires: 60 <-------------> [Feb 22 17:24:17] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:24:17] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK7f8eb05f (58) [Feb 22 17:24:17] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: (40) [Feb 22 17:24:17] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (38) [Feb 22 17:24:17] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 000ff78d-ebb20004-75d3ddb3-71096c5c@192.168.1.165 (58) [Feb 22 17:24:17] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Date: Thu, 22 Feb 2007 22:25:51 GMT (35) [Feb 22 17:24:17] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: CSeq: 49692 REGISTER (20) [Feb 22 17:24:17] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: CSCO/7 (18) [Feb 22 17:24:17] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Contact: (37) [Feb 22 17:24:17] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:24:17] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Expires: 60 (11) [Feb 22 17:24:17] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: (0) [Feb 22 17:24:17] --- (11 headers 0 lines) --- [Feb 22 17:24:17] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 000ff78d-ebb20004-75d3ddb3-71096c5c@192.168.1.165 - REGISTER (No RTP) [Feb 22 17:24:17] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:24:17] Using latest REGISTER request as basis request [Feb 22 17:24:17] Sending to 192.168.1.165 : 5060 (no NAT) [Feb 22 17:24:17] <--- 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=z9hG4bK7f8eb05f;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20004-75d3ddb3-71096c5c@192.168.1.165 CSeq: 49692 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:24:17] <--- 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=z9hG4bK7f8eb05f;received=192.168.1.165 From: To: ;tag=as181f8b85 Call-ID: 000ff78d-ebb20004-75d3ddb3-71096c5c@192.168.1.165 CSeq: 49692 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="621adf36" Content-Length: 0 <------------> [Feb 22 17:24:17] Scheduling destruction of SIP dialog '000ff78d-ebb20004-75d3ddb3-71096c5c@192.168.1.165' in 32000 ms (Method: REGISTER) [Feb 22 17:24:17] <--- 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=z9hG4bK3351e5b1 From: To: Call-ID: 000ff78d-ebb20004-75d3ddb3-71096c5c@192.168.1.165 Date: Thu, 22 Feb 2007 22:25:51 GMT CSeq: 49693 REGISTER User-Agent: CSCO/7 Contact: Authorization: Digest username="512",realm="asterisk",uri="sip:24.123.23.170",response="5346fdc9c17562453b286d6cbd1ccb55",nonce="621adf36",algorithm=MD5 Content-Length: 0 Expires: 60 <-------------> [Feb 22 17:24:17] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:24:17] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK3351e5b1 (58) [Feb 22 17:24:17] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: (40) [Feb 22 17:24:17] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (38) [Feb 22 17:24:17] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 000ff78d-ebb20004-75d3ddb3-71096c5c@192.168.1.165 (58) [Feb 22 17:24:17] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Date: Thu, 22 Feb 2007 22:25:51 GMT (35) [Feb 22 17:24:17] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: CSeq: 49693 REGISTER (20) [Feb 22 17:24:17] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: CSCO/7 (18) [Feb 22 17:24:17] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Contact: (37) [Feb 22 17:24:17] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Authorization: Digest username="512",realm="asterisk",uri="sip:24.123.23.170",response="5346fdc9c17562453b286d6cbd1ccb55",nonce="621adf36",algorithm=MD5 (152) [Feb 22 17:24:17] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Content-Length: 0 (17) [Feb 22 17:24:17] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Expires: 60 (11) [Feb 22 17:24:17] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: (0) [Feb 22 17:24:17] --- (12 headers 0 lines) --- [Feb 22 17:24:17] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:24:17] Using latest REGISTER request as basis request [Feb 22 17:24:17] Sending to 192.168.1.165 : 5060 (no NAT) [Feb 22 17:24:17] <--- 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=z9hG4bK3351e5b1;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20004-75d3ddb3-71096c5c@192.168.1.165 CSeq: 49693 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:24:17] <--- 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=z9hG4bK3351e5b1;received=192.168.1.165 From: To: ;tag=as181f8b85 Call-ID: 000ff78d-ebb20004-75d3ddb3-71096c5c@192.168.1.165 CSeq: 49693 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Thu, 22 Feb 2007 22:24:17 GMT Content-Length: 0 <------------> [Feb 22 17:24:17] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/512 [Feb 22 17:24:17] Scheduling destruction of SIP dialog '000ff78d-ebb20004-75d3ddb3-71096c5c@192.168.1.165' in 32000 ms (Method: REGISTER) [Feb 22 17:24:17] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 512 [Feb 22 17:24:17] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 512 [Feb 22 17:24:17] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/512 - state 1 (Not in use) [Feb 22 17:24:17] DEBUG[14835]: app_queue.c:546 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. [Feb 22 17:24:19] NOTICE[14767]: chan_sip.c:7055 sip_reregister: -- Re-registration for 3173241052@sip.broadvoice.com [Feb 22 17:24:19] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 42fcb9a2205a226e1657915a20f1c05a@24.123.23.170 - REGISTER (No RTP) [Feb 22 17:24:19] DEBUG[14767]: chan_sip.c:7214 transmit_register: Scheduled a registration timeout for sip.broadvoice.com id #194 [Feb 22 17:24:19] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:sip.broadvoice.com SIP/2.0 (39) [Feb 22 17:24:19] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK10832d70;rport (64) [Feb 22 17:24:19] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: ;tag=as19c54767 (56) [Feb 22 17:24:19] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (39) [Feb 22 17:24:19] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 42fcb9a2205a226e1657915a20f1c05a@24.123.23.170 (55) [Feb 22 17:24:19] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: CSeq: 107 REGISTER (18) [Feb 22 17:24:19] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Feb 22 17:24:19] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: Max-Forwards: 70 (16) [Feb 22 17:24:19] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Expires: 120 (12) [Feb 22 17:24:19] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Contact: (39) [Feb 22 17:24:19] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Event: registration (19) [Feb 22 17:24:19] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Content-Length: 0 (17) [Feb 22 17:24:19] REGISTER 12 headers, 0 lines [Feb 22 17:24:19] REGISTER attempt 1 to 3173241052@sip.broadvoice.com [Feb 22 17:24:19] 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=z9hG4bK10832d70;rport From: ;tag=as19c54767 To: Call-ID: 42fcb9a2205a226e1657915a20f1c05a@24.123.23.170 CSeq: 107 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: Event: registration Content-Length: 0 --- [Feb 22 17:24:19] DEBUG[14767]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #195 [Feb 22 17:24:19] <--- SIP read from 147.135.12.128:5060 ---> SIP/2.0 200 OK Call-ID: 42fcb9a2205a226e1657915a20f1c05a@24.123.23.170 CSeq: 107 REGISTER From: ;tag=as19c54767 To: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK10832d70 Contact: Expires: 30 Event: registration Content-Length: 0 <-------------> [Feb 22 17:24:19] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 22 17:24:19] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Call-ID: 42fcb9a2205a226e1657915a20f1c05a@24.123.23.170 (55) [Feb 22 17:24:19] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: CSeq: 107 REGISTER (18) [Feb 22 17:24:19] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: From: ;tag=as19c54767 (56) [Feb 22 17:24:19] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: To: (39) [Feb 22 17:24:19] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK10832d70 (58) [Feb 22 17:24:19] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: (39) [Feb 22 17:24:19] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: Expires: 30 (11) [Feb 22 17:24:19] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Event: registration (19) [Feb 22 17:24:19] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (20) [Feb 22 17:24:19] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: (0) [Feb 22 17:24:19] --- (10 headers 0 lines) --- [Feb 22 17:24:19] DEBUG[14767]: chan_sip.c:2061 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #195 [Feb 22 17:24:19] DEBUG[14767]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '42fcb9a2205a226e1657915a20f1c05a@24.123.23.170' of Request 107: Match Not Found [Feb 22 17:24:19] DEBUG[14767]: chan_sip.c:11906 handle_response_register: Registration successful [Feb 22 17:24:19] DEBUG[14767]: chan_sip.c:11909 handle_response_register: Cancelling timeout 194 [Feb 22 17:24:19] Scheduling destruction of SIP dialog '42fcb9a2205a226e1657915a20f1c05a@24.123.23.170' in 32000 ms (Method: REGISTER) [Feb 22 17:24:19] NOTICE[14767]: chan_sip.c:11961 handle_response_register: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 23 s) [Feb 22 17:24:20] <--- SIP read from 192.168.1.93:5060 ---> INVITE sip:550@24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKz396a89a1f449c1571d264c89a7298159 Call-ID: 1ca0c9900034b45264c7ee1a67dc712c@24.123.23.170 CSeq: 487057 INVITE From: 528 ;tag=177014cec8654efa2f4863c390adf2a0 To: ;tag=as239b687a Contact: 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: 242 v=0 o=- 14736 345899 IN IP4 192.168.1.93 s=session c=IN IP4 0.0.0.0 t=0 0 m=audio 30032 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 - - - - a=ptime:20 a=sendonly <-------------> [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: INVITE sip:550@24.123.23.170 SIP/2.0 (36) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKz396a89a1f449c1571d264c89a7298159 (82) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: Call-ID: 1ca0c9900034b45264c7ee1a67dc712c@24.123.23.170 (55) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: CSeq: 487057 INVITE (19) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: 528 ;tag=177014cec8654efa2f4863c390adf2a0 (74) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: To: ;tag=as239b687a (42) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: (36) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: Content-Type: application/sdp (29) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Max-Forwards: 70 (16) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: Content-Length: 242 (19) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 13: (0) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: v=0 (3) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: o=- 14736 345899 IN IP4 192.168.1.93 (36) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: s=session (9) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: c=IN IP4 0.0.0.0 (16) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: t=0 0 (5) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: m=audio 30032 RTP/AVP 0 101 (27) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=fmtp:101 0-15,16,17,18 (24) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=silenceSupp:off - - - - (25) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=ptime:20 (10) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=sendonly (10) [Feb 22 17:24:20] --- (13 headers 12 lines) --- [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:1662 parse_sip_options: Begin: parsing SIP "Supported: sip-cc, sip-cc-01, replaces, timer" [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:1670 parse_sip_options: Found SIP option: -sip-cc- [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:1684 parse_sip_options: Found no match for SIP option: sip-cc (Please file bug report!) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:1670 parse_sip_options: Found SIP option: -sip-cc-01- [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:1684 parse_sip_options: Found no match for SIP option: sip-cc-01 (Please file bug report!) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:1670 parse_sip_options: Found SIP option: -replaces- [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:1676 parse_sip_options: Matched SIP option: replaces [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:1670 parse_sip_options: Found SIP option: -timer- [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:1676 parse_sip_options: Matched SIP option: timer [Feb 22 17:24:20] Sending to 192.168.1.93 : 5060 (no NAT) [Feb 22 17:24:20] Found RTP audio format 0 [Feb 22 17:24:20] Found RTP audio format 101 [Feb 22 17:24:20] Peer audio RTP is at port 0.0.0.0:30032 [Feb 22 17:24:20] Found description format PCMU for ID 0 [Feb 22 17:24:20] Found description format telephone-event for ID 101 [Feb 22 17:24:20] Got unsupported a:fmtp in SDP offer [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:5068 process_sdp: T38 state changed to 0 on channel SIP/528-09e0cbb8 [Feb 22 17:24:20] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Feb 22 17:24:20] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Feb 22 17:24:20] Peer audio RTP is at port 0.0.0.0:30032 [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:5145 process_sdp: We're settling with these formats: 0x4 (ulaw) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:5152 process_sdp: We have an owner, now see if we need to change this call [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:13288 handle_request_invite: Got a SIP re-invite for call 1ca0c9900034b45264c7ee1a67dc712c@24.123.23.170 [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:13383 handle_request_invite: SIP/528-09e0cbb8: This call is UP.... [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:6354 transmit_response_with_sdp: Setting framing from config on incoming call [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:6122 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: True [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:6123 add_sdp: ** Our prefcodec: 0x4 (ulaw) [Feb 22 17:24:20] Audio is at 24.123.23.170 port 19576 [Feb 22 17:24:20] Adding codec 0x4 (ulaw) to SDP [Feb 22 17:24:20] Adding non-codec 0x1 (telephone-event) to SDP [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:6254 add_sdp: -- Done with adding codecs to SDP [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:6299 add_sdp: Done building SDP. Settling with this capability: 0x4 (ulaw) [Feb 22 17:24:20] <--- Reliably 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=z9hG4bKz396a89a1f449c1571d264c89a7298159;received=192.168.1.93 From: 528 ;tag=177014cec8654efa2f4863c390adf2a0 To: ;tag=as239b687a Call-ID: 1ca0c9900034b45264c7ee1a67dc712c@24.123.23.170 CSeq: 487057 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 240 v=0 o=root 14736 14737 IN IP4 192.168.1.91 s=session c=IN IP4 192.168.1.91 t=0 0 m=audio 10010 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #198 [Feb 22 17:24:20] DEBUG[14828]: channel.c:2731 set_format: Set channel SIP/550-09e07bc8 to write format slin [Feb 22 17:24:20] -- Started music on hold, class 'default', on channel 'SIP/550-09e07bc8' [Feb 22 17:24:20] DEBUG[14828]: channel.c:1908 ast_settimeout: Scheduling timer at 160 sample intervals [Feb 22 17:24:20] DEBUG[14828]: rtp.c:2783 bridge_native_loop: Oooh, 'SIP/528-09e0cbb8' changed end address to 0.0.0.0:0 (format 4) [Feb 22 17:24:20] DEBUG[14828]: rtp.c:2785 bridge_native_loop: Oooh, 'SIP/528-09e0cbb8' changed end vaddress to 0.0.0.0:0 (format 4) [Feb 22 17:24:20] DEBUG[14828]: rtp.c:2787 bridge_native_loop: Oooh, 'SIP/528-09e0cbb8' was 192.168.1.93:30032/(format 4) [Feb 22 17:24:20] DEBUG[14828]: rtp.c:2789 bridge_native_loop: Oooh, 'SIP/528-09e0cbb8' was 0.0.0.0:0/(format 4) [Feb 22 17:24:20] DEBUG[14828]: chan_sip.c:16704 sip_set_rtp_peer: Sending reinvite on SIP '1735147470@192.168.1.91' - It's audio soon redirected to IP 24.123.23.170 [Feb 22 17:24:20] DEBUG[14828]: chan_sip.c:5576 reqprep: Strict routing enforced for session 1735147470@192.168.1.91 [Feb 22 17:24:20] set_destination: Parsing for address/port to send to [Feb 22 17:24:20] set_destination: set destination to 192.168.1.91, port 5060 [Feb 22 17:24:20] DEBUG[14828]: chan_sip.c:6122 add_sdp: ** Our capability: 0xc (ulaw|alaw) Video flag: True [Feb 22 17:24:20] DEBUG[14828]: chan_sip.c:6123 add_sdp: ** Our prefcodec: 0x0 (nothing) [Feb 22 17:24:20] Audio is at 24.123.23.170 port 15940 [Feb 22 17:24:20] Adding codec 0x4 (ulaw) to SDP [Feb 22 17:24:20] Adding codec 0x8 (alaw) to SDP [Feb 22 17:24:20] Adding non-codec 0x1 (telephone-event) to SDP [Feb 22 17:24:20] DEBUG[14828]: chan_sip.c:6254 add_sdp: -- Done with adding codecs to SDP [Feb 22 17:24:20] DEBUG[14828]: chan_sip.c:6299 add_sdp: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Feb 22 17:24:20] DEBUG[14828]: chan_sip.c:1603 initialize_initreq: Initializing already initialized SIP dialog 1735147470@192.168.1.91 (presumably reinvite) [Feb 22 17:24:20] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 0: INVITE sip:550@192.168.1.91:5060 SIP/2.0 (40) [Feb 22 17:24:20] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK7d6ba559;rport (64) [Feb 22 17:24:20] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 2: From: ;tag=as2b86d5e2 (49) [Feb 22 17:24:20] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 3: To: "Display Name" ;tag=2772323005 (62) [Feb 22 17:24:20] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 4: Contact: (32) [Feb 22 17:24:20] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 5: Call-ID: 1735147470@192.168.1.91 (32) [Feb 22 17:24:20] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 6: CSeq: 102 INVITE (16) [Feb 22 17:24:20] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Feb 22 17:24:20] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 22 17:24:20] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Feb 22 17:24:20] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 10: Supported: replaces (19) [Feb 22 17:24:20] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Feb 22 17:24:20] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 12: Content-Type: application/sdp (29) [Feb 22 17:24:20] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 13: Content-Length: 266 (19) [Feb 22 17:24:20] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 14: (0) [Feb 22 17:24:20] DEBUG[14828]: chan_sip.c:4559 parse_request: Line: v=0 (3) [Feb 22 17:24:20] DEBUG[14828]: chan_sip.c:4559 parse_request: Line: o=root 14736 14737 IN IP4 24.123.23.170 (39) [Feb 22 17:24:20] DEBUG[14828]: chan_sip.c:4559 parse_request: Line: s=session (9) [Feb 22 17:24:20] DEBUG[14828]: chan_sip.c:4559 parse_request: Line: c=IN IP4 24.123.23.170 (22) [Feb 22 17:24:20] DEBUG[14828]: chan_sip.c:4559 parse_request: Line: t=0 0 (5) [Feb 22 17:24:20] DEBUG[14828]: chan_sip.c:4559 parse_request: Line: m=audio 15940 RTP/AVP 0 8 101 (29) [Feb 22 17:24:20] DEBUG[14828]: chan_sip.c:4559 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Feb 22 17:24:20] DEBUG[14828]: chan_sip.c:4559 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Feb 22 17:24:20] DEBUG[14828]: chan_sip.c:4559 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Feb 22 17:24:20] DEBUG[14828]: chan_sip.c:4559 parse_request: Line: a=fmtp:101 0-16 (15) [Feb 22 17:24:20] DEBUG[14828]: chan_sip.c:4559 parse_request: Line: a=silenceSupp:off - - - - (25) [Feb 22 17:24:20] DEBUG[14828]: chan_sip.c:4559 parse_request: Line: a=ptime:20 (10) [Feb 22 17:24:20] DEBUG[14828]: chan_sip.c:4559 parse_request: Line: a=sendrecv (10) [Feb 22 17:24:20] Reliably Transmitting (NAT) to 192.168.1.91:5060: INVITE sip:550@192.168.1.91:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK7d6ba559;rport From: ;tag=as2b86d5e2 To: "Display Name" ;tag=2772323005 Contact: Call-ID: 1735147470@192.168.1.91 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 266 v=0 o=root 14736 14737 IN IP4 24.123.23.170 s=session c=IN IP4 24.123.23.170 t=0 0 m=audio 15940 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 - - - - a=ptime:20 a=sendrecv --- [Feb 22 17:24:20] DEBUG[14828]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #199 [Feb 22 17:24:20] DEBUG[14828]: channel.c:1689 generator_force: Auto-deactivating generator [Feb 22 17:24:20] DEBUG[14828]: channel.c:2731 set_format: Set channel SIP/550-09e07bc8 to write format ulaw [Feb 22 17:24:20] -- Stopped music on hold on SIP/550-09e07bc8 [Feb 22 17:24:20] DEBUG[14828]: channel.c:1908 ast_settimeout: Scheduling timer at 0 sample intervals [Feb 22 17:24:20] <--- SIP read from 192.168.1.93:5060 ---> ACK sip:550@24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKq201ef56e1756cbee92ede212e838fd16 CSeq: 487057 ACK To: ;tag=as239b687a Call-ID: 1ca0c9900034b45264c7ee1a67dc712c@24.123.23.170 From: 528 ;tag=177014cec8654efa2f4863c390adf2a0 User-Agent: Uniden SIP Phone p2 Ver BS4.77 <-------------> [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: ACK sip:550@24.123.23.170 SIP/2.0 (33) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKq201ef56e1756cbee92ede212e838fd16 (82) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: CSeq: 487057 ACK (16) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: ;tag=as239b687a (42) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 1ca0c9900034b45264c7ee1a67dc712c@24.123.23.170 (55) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: From: 528 ;tag=177014cec8654efa2f4863c390adf2a0 (74) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: (0) [Feb 22 17:24:20] --- (7 headers 0 lines) --- [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received ACK (6) - Command in SIP ACK [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:2061 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #198 [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '1ca0c9900034b45264c7ee1a67dc712c@24.123.23.170' of Response 487057: Match Not Found [Feb 22 17:24:20] <--- 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=z9hG4bK71cb8f33 From: To: Call-ID: 000ff78d-ebb20006-6007ecd7-0ae287ec@192.168.1.165 Date: Thu, 22 Feb 2007 22:25:54 GMT CSeq: 49733 REGISTER User-Agent: CSCO/7 Contact: Content-Length: 0 Expires: 60 <-------------> [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK71cb8f33 (58) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: (40) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (38) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 000ff78d-ebb20006-6007ecd7-0ae287ec@192.168.1.165 (58) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Date: Thu, 22 Feb 2007 22:25:54 GMT (35) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: CSeq: 49733 REGISTER (20) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: CSCO/7 (18) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Contact: (37) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Expires: 60 (11) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: (0) [Feb 22 17:24:20] --- (11 headers 0 lines) --- [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 000ff78d-ebb20006-6007ecd7-0ae287ec@192.168.1.165 - REGISTER (No RTP) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:24:20] Using latest REGISTER request as basis request [Feb 22 17:24:20] Sending to 192.168.1.165 : 5060 (no NAT) [Feb 22 17:24:20] <--- 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=z9hG4bK71cb8f33;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20006-6007ecd7-0ae287ec@192.168.1.165 CSeq: 49733 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:24:20] <--- 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=z9hG4bK71cb8f33;received=192.168.1.165 From: To: ;tag=as7ff06f2d Call-ID: 000ff78d-ebb20006-6007ecd7-0ae287ec@192.168.1.165 CSeq: 49733 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0b4cbfe1" Content-Length: 0 <------------> [Feb 22 17:24:20] Scheduling destruction of SIP dialog '000ff78d-ebb20006-6007ecd7-0ae287ec@192.168.1.165' in 32000 ms (Method: REGISTER) [Feb 22 17:24:20] <--- SIP read from 192.168.1.93:5060 ---> INVITE sip:526@24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKm7ef1a99297826597cbf65730292cc528 Call-ID: 524091ef199964d13508e0184586dc1c@24.123.23.170 CSeq: 440469 INVITE From: 528 ;tag=81c5c0cdcd26df8f2948fc3e1971f0eb To: ;tag=as449238ce Contact: 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: 267 v=0 o=- 710591507 70058 IN IP4 192.168.1.93 s=- c=IN IP4 192.168.1.93 t=0 0 m=audio 30030 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 <-------------> [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: INVITE sip:526@24.123.23.170 SIP/2.0 (36) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKm7ef1a99297826597cbf65730292cc528 (82) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: Call-ID: 524091ef199964d13508e0184586dc1c@24.123.23.170 (55) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: CSeq: 440469 INVITE (19) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: 528 ;tag=81c5c0cdcd26df8f2948fc3e1971f0eb (74) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: To: ;tag=as449238ce (42) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: (36) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: Content-Type: application/sdp (29) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Max-Forwards: 70 (16) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: Content-Length: 267 (19) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 13: (0) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: v=0 (3) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: o=- 710591507 70058 IN IP4 192.168.1.93 (39) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: s=- (3) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: c=IN IP4 192.168.1.93 (21) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: t=0 0 (5) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: m=audio 30030 RTP/AVP 0 8 18 101 (32) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=fmtp:101 0-15,16,17,18 (24) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=sendrecv (10) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=ptime:20 (10) [Feb 22 17:24:20] --- (13 headers 13 lines) --- [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Feb 22 17:24:20] Sending to 192.168.1.93 : 5060 (no NAT) [Feb 22 17:24:20] Found RTP audio format 0 [Feb 22 17:24:20] Found RTP audio format 8 [Feb 22 17:24:20] Found RTP audio format 18 [Feb 22 17:24:20] Found RTP audio format 101 [Feb 22 17:24:20] Peer audio RTP is at port 192.168.1.93:30030 [Feb 22 17:24:20] Found description format PCMU for ID 0 [Feb 22 17:24:20] Found description format PCMA for ID 8 [Feb 22 17:24:20] Found description format G729 for ID 18 [Feb 22 17:24:20] Found description format telephone-event for ID 101 [Feb 22 17:24:20] Got unsupported a:fmtp in SDP offer [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:5068 process_sdp: T38 state changed to 0 on channel SIP/528-09e02620 [Feb 22 17:24:20] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) [Feb 22 17:24:20] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Feb 22 17:24:20] Peer audio RTP is at port 192.168.1.93:30030 [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:5145 process_sdp: We're settling with these formats: 0xc (ulaw|alaw) [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:5152 process_sdp: We have an owner, now see if we need to change this call [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:13288 handle_request_invite: Got a SIP re-invite for call 524091ef199964d13508e0184586dc1c@24.123.23.170 [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:13383 handle_request_invite: SIP/528-09e02620: This call is UP.... [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:6354 transmit_response_with_sdp: Setting framing from config on incoming call [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:6122 add_sdp: ** Our capability: 0xc (ulaw|alaw) Video flag: True [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:6123 add_sdp: ** Our prefcodec: 0x4 (ulaw) [Feb 22 17:24:20] Audio is at 24.123.23.170 port 16212 [Feb 22 17:24:20] Adding codec 0x4 (ulaw) to SDP [Feb 22 17:24:20] Adding codec 0x8 (alaw) to SDP [Feb 22 17:24:20] Adding non-codec 0x1 (telephone-event) to SDP [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:6254 add_sdp: -- Done with adding codecs to SDP [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:6299 add_sdp: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Feb 22 17:24:20] <--- Reliably 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=z9hG4bKm7ef1a99297826597cbf65730292cc528;received=192.168.1.93 From: 528 ;tag=81c5c0cdcd26df8f2948fc3e1971f0eb To: ;tag=as449238ce Call-ID: 524091ef199964d13508e0184586dc1c@24.123.23.170 CSeq: 440469 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 264 v=0 o=root 14736 14738 IN IP4 192.168.1.66 s=session c=IN IP4 192.168.1.66 t=0 0 m=audio 30006 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 - - - - a=ptime:20 a=sendrecv <------------> [Feb 22 17:24:20] DEBUG[14767]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #201 [Feb 22 17:24:20] DEBUG[14821]: rtp.c:2783 bridge_native_loop: Oooh, 'SIP/528-09e02620' changed end address to 192.168.1.93:30030 (format 268) [Feb 22 17:24:20] DEBUG[14821]: rtp.c:2785 bridge_native_loop: Oooh, 'SIP/528-09e02620' changed end vaddress to 0.0.0.0:0 (format 268) [Feb 22 17:24:20] DEBUG[14821]: rtp.c:2787 bridge_native_loop: Oooh, 'SIP/528-09e02620' was 0.0.0.0:0/(format 4) [Feb 22 17:24:20] DEBUG[14821]: rtp.c:2789 bridge_native_loop: Oooh, 'SIP/528-09e02620' was 0.0.0.0:0/(format 4) [Feb 22 17:24:20] DEBUG[14821]: chan_sip.c:16704 sip_set_rtp_peer: Sending reinvite on SIP '903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170' - It's audio soon redirected to IP 192.168.1.93 [Feb 22 17:24:20] DEBUG[14821]: chan_sip.c:5576 reqprep: Strict routing enforced for session 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 [Feb 22 17:24:20] set_destination: Parsing for address/port to send to [Feb 22 17:24:20] set_destination: set destination to 192.168.1.66, port 5060 [Feb 22 17:24:20] DEBUG[14821]: chan_sip.c:6122 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: True [Feb 22 17:24:20] DEBUG[14821]: chan_sip.c:6123 add_sdp: ** Our prefcodec: 0x0 (nothing) [Feb 22 17:24:20] Audio is at 24.123.23.170 port 15904 [Feb 22 17:24:20] Adding codec 0x4 (ulaw) to SDP [Feb 22 17:24:20] Adding non-codec 0x1 (telephone-event) to SDP [Feb 22 17:24:20] DEBUG[14821]: chan_sip.c:6254 add_sdp: -- Done with adding codecs to SDP [Feb 22 17:24:20] DEBUG[14821]: chan_sip.c:6299 add_sdp: Done building SDP. Settling with this capability: 0x4 (ulaw) [Feb 22 17:24:20] DEBUG[14821]: chan_sip.c:1603 initialize_initreq: Initializing already initialized SIP dialog 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 (presumably reinvite) [Feb 22 17:24:20] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 0: INVITE sip:526@192.168.1.66:5060 SIP/2.0 (40) [Feb 22 17:24:20] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK041a4be5 (58) [Feb 22 17:24:20] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 2: From: ;tag=as056fdf38 (44) [Feb 22 17:24:20] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 3: To: 526 ;tag=48106fce45713fff4b942431e90afa0e (68) [Feb 22 17:24:20] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 4: Contact: (32) [Feb 22 17:24:20] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 5: Call-ID: 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 (68) [Feb 22 17:24:20] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 6: CSeq: 103 INVITE (16) [Feb 22 17:24:20] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Feb 22 17:24:20] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 22 17:24:20] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Feb 22 17:24:20] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 10: Supported: replaces (19) [Feb 22 17:24:20] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Feb 22 17:24:20] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 12: Content-Type: application/sdp (29) [Feb 22 17:24:20] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 13: Content-Length: 240 (19) [Feb 22 17:24:20] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 14: (0) [Feb 22 17:24:20] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: v=0 (3) [Feb 22 17:24:20] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: o=root 14736 14738 IN IP4 192.168.1.93 (38) [Feb 22 17:24:20] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: s=session (9) [Feb 22 17:24:20] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: c=IN IP4 192.168.1.93 (21) [Feb 22 17:24:20] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: t=0 0 (5) [Feb 22 17:24:20] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: m=audio 30030 RTP/AVP 0 101 (27) [Feb 22 17:24:20] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Feb 22 17:24:20] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Feb 22 17:24:20] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: a=fmtp:101 0-16 (15) [Feb 22 17:24:20] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: a=silenceSupp:off - - - - (25) [Feb 22 17:24:20] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: a=ptime:20 (10) [Feb 22 17:24:20] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: a=sendrecv (10) [Feb 22 17:24:20] Reliably Transmitting (no NAT) to 192.168.1.66:5060: INVITE sip:526@192.168.1.66:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK041a4be5 From: ;tag=as056fdf38 To: 526 ;tag=48106fce45713fff4b942431e90afa0e Contact: Call-ID: 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 240 v=0 o=root 14736 14738 IN IP4 192.168.1.93 s=session c=IN IP4 192.168.1.93 t=0 0 m=audio 30030 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Feb 22 17:24:20] DEBUG[14821]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #202 [Feb 22 17:24:20] DEBUG[14821]: rtp.c:2647 ast_rtp_write: Ooh, format changed from unknown to ulaw [Feb 22 17:24:20] DEBUG[14821]: rtp.c:2664 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Feb 22 17:24:21] <--- SIP read from 192.168.1.66:5060 ---> SIP/2.0 200 OK To: 526 ;tag=48106fce45713fff4b942431e90afa0e Contact: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK041a4be5 From: ;tag=as056fdf38 Call-ID: 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 CSeq: 103 INVITE Content-Type: application/sdp Content-Length: 247 Supported: sip-cc, sip-cc-01, replaces, timer Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK v=0 o=- 14736 163217 IN IP4 192.168.1.66 s=session c=IN IP4 192.168.1.66 t=0 0 m=audio 30006 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 - - - - a=ptime:20 a=sendrecv <-------------> [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: To: 526 ;tag=48106fce45713fff4b942431e90afa0e (68) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: Contact: (36) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK041a4be5 (58) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: ;tag=as056fdf38 (44) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Call-ID: 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 (68) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: CSeq: 103 INVITE (16) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: Content-Type: application/sdp (29) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Content-Length: 247 (19) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: (0) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: v=0 (3) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: o=- 14736 163217 IN IP4 192.168.1.66 (36) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: s=session (9) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: c=IN IP4 192.168.1.66 (21) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: t=0 0 (5) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: m=audio 30006 RTP/AVP 0 101 (27) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=fmtp:101 0-15,16,17,18 (24) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=silenceSupp:off - - - - (25) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=ptime:20 (10) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=sendrecv (10) [Feb 22 17:24:21] --- (11 headers 12 lines) --- [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:2053 __sip_ack: Acked pending invite 103 [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:2061 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #202 [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170' of Request 103: Match Not Found [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:11529 handle_response_invite: SIP response 200 to RE-invite on outgoing call 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 [Feb 22 17:24:21] Found RTP audio format 0 [Feb 22 17:24:21] Found RTP audio format 101 [Feb 22 17:24:21] Peer audio RTP is at port 192.168.1.66:30006 [Feb 22 17:24:21] Found description format PCMU for ID 0 [Feb 22 17:24:21] Found description format telephone-event for ID 101 [Feb 22 17:24:21] Got unsupported a:fmtp in SDP offer [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:5068 process_sdp: T38 state changed to 0 on channel SIP/526-09df30e0 [Feb 22 17:24:21] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Feb 22 17:24:21] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Feb 22 17:24:21] Peer audio RTP is at port 192.168.1.66:30006 [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:5145 process_sdp: We're settling with these formats: 0x4 (ulaw) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:5152 process_sdp: We have an owner, now see if we need to change this call [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:2978 update_call_counter: Updating call counter for outgoing call [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:7830 build_route: build_route: Retaining previous route: [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:11662 handle_response_invite: Strange... The other side of the bridge does not have a udptl struct [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:11667 handle_response_invite: T38 state changed to 0 on channel SIP [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:11670 handle_response_invite: T38 state changed to 0 on channel SIP/526-09df30e0 [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:5576 reqprep: Strict routing enforced for session 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 [Feb 22 17:24:21] set_destination: Parsing for address/port to send to [Feb 22 17:24:21] set_destination: set destination to 192.168.1.66, port 5060 [Feb 22 17:24:21] Transmitting (no NAT) to 192.168.1.66:5060: ACK sip:526@192.168.1.66:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK2dd090bd From: ;tag=as056fdf38 To: 526 ;tag=48106fce45713fff4b942431e90afa0e Contact: Call-ID: 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Feb 22 17:24:21] <--- 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=z9hG4bK4d8e1c65 From: To: Call-ID: 000ff78d-ebb20006-6007ecd7-0ae287ec@192.168.1.165 Date: Thu, 22 Feb 2007 22:25:54 GMT CSeq: 49734 REGISTER User-Agent: CSCO/7 Contact: Authorization: Digest username="514",realm="asterisk",uri="sip:24.123.23.170",response="e5c1d2f5e5413d8ddf669b65e021332f",nonce="0b4cbfe1",algorithm=MD5 Content-Length: 0 Expires: 60 <-------------> [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK4d8e1c65 (58) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: (40) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (38) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 000ff78d-ebb20006-6007ecd7-0ae287ec@192.168.1.165 (58) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Date: Thu, 22 Feb 2007 22:25:54 GMT (35) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: CSeq: 49734 REGISTER (20) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: CSCO/7 (18) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Contact: (37) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Authorization: Digest username="514",realm="asterisk",uri="sip:24.123.23.170",response="e5c1d2f5e5413d8ddf669b65e021332f",nonce="0b4cbfe1",algorithm=MD5 (152) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Content-Length: 0 (17) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Expires: 60 (11) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: (0) [Feb 22 17:24:21] --- (12 headers 0 lines) --- [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:24:21] Using latest REGISTER request as basis request [Feb 22 17:24:21] Sending to 192.168.1.165 : 5060 (no NAT) [Feb 22 17:24:21] <--- 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=z9hG4bK4d8e1c65;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20006-6007ecd7-0ae287ec@192.168.1.165 CSeq: 49734 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:24:21] <--- 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=z9hG4bK4d8e1c65;received=192.168.1.165 From: To: ;tag=as7ff06f2d Call-ID: 000ff78d-ebb20006-6007ecd7-0ae287ec@192.168.1.165 CSeq: 49734 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Thu, 22 Feb 2007 22:24:21 GMT Content-Length: 0 <------------> [Feb 22 17:24:21] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/514 [Feb 22 17:24:21] Scheduling destruction of SIP dialog '000ff78d-ebb20006-6007ecd7-0ae287ec@192.168.1.165' in 32000 ms (Method: REGISTER) [Feb 22 17:24:21] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 514 [Feb 22 17:24:21] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 514 [Feb 22 17:24:21] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/514 - state 1 (Not in use) [Feb 22 17:24:21] DEBUG[14836]: app_queue.c:546 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. [Feb 22 17:24:21] <--- 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=z9hG4bKcbe08a630aa8205113fbca757f9ff39a5 Call-ID: 55ae664767e0@24.123.23.170 CSeq: 73554 REGISTER From: ;tag=4c2fecd4d2340bbbaadd498c30aca938 To: Contact: ;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="41b6d80e", algorithm=MD5, uri="sip:24.123.23.170", username="523", response="c62d8a32110b4fee2091d8efe1158013" <-------------> [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKcbe08a630aa8205113fbca757f9ff39a5 (82) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: Call-ID: 55ae664767e0@24.123.23.170 (35) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: CSeq: 73554 REGISTER (20) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: ;tag=4c2fecd4d2340bbbaadd498c30aca938 (66) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: To: (27) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="41b6d80e", algorithm=MD5, uri="sip:24.123.23.170", username="523", response="c62d8a32110b4fee2091d8efe1158013" (167) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 13: (0) [Feb 22 17:24:21] --- (13 headers 0 lines) --- [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 55ae664767e0@24.123.23.170 - REGISTER (No RTP) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:24:21] Using latest REGISTER request as basis request [Feb 22 17:24:21] Sending to 192.168.1.85 : 5060 (no NAT) [Feb 22 17:24:21] <--- 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=z9hG4bKcbe08a630aa8205113fbca757f9ff39a5;received=192.168.1.85 From: ;tag=4c2fecd4d2340bbbaadd498c30aca938 To: Call-ID: 55ae664767e0@24.123.23.170 CSeq: 73554 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:24:21] <--- 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=z9hG4bKcbe08a630aa8205113fbca757f9ff39a5;received=192.168.1.85 From: ;tag=4c2fecd4d2340bbbaadd498c30aca938 To: ;tag=as49f11a59 Call-ID: 55ae664767e0@24.123.23.170 CSeq: 73554 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="23dc9c01" Content-Length: 0 <------------> [Feb 22 17:24:21] Scheduling destruction of SIP dialog '55ae664767e0@24.123.23.170' in 32000 ms (Method: REGISTER) [Feb 22 17:24:21] <--- SIP read from 192.168.1.93:5060 ---> INVITE sip:526@24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKm7ef1a99297826597cbf65730292cc528 Call-ID: 524091ef199964d13508e0184586dc1c@24.123.23.170 CSeq: 440469 INVITE From: 528 ;tag=81c5c0cdcd26df8f2948fc3e1971f0eb To: ;tag=as449238ce Contact: 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: 267 v=0 o=- 710591507 70058 IN IP4 192.168.1.93 s=- c=IN IP4 192.168.1.93 t=0 0 m=audio 30030 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 <-------------> [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: INVITE sip:526@24.123.23.170 SIP/2.0 (36) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKm7ef1a99297826597cbf65730292cc528 (82) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: Call-ID: 524091ef199964d13508e0184586dc1c@24.123.23.170 (55) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: CSeq: 440469 INVITE (19) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: 528 ;tag=81c5c0cdcd26df8f2948fc3e1971f0eb (74) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: To: ;tag=as449238ce (42) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: (36) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: Content-Type: application/sdp (29) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Max-Forwards: 70 (16) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: Content-Length: 267 (19) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 13: (0) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: v=0 (3) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: o=- 710591507 70058 IN IP4 192.168.1.93 (39) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: s=- (3) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: c=IN IP4 192.168.1.93 (21) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: t=0 0 (5) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: m=audio 30030 RTP/AVP 0 8 18 101 (32) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=fmtp:101 0-15,16,17,18 (24) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=sendrecv (10) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=ptime:20 (10) [Feb 22 17:24:21] --- (13 headers 13 lines) --- [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:14472 handle_request: Ignoring SIP message because of retransmit (INVITE Seqno 440469, ours 440469) [Feb 22 17:24:21] Ignoring this INVITE request [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:13290 handle_request_invite: Got a SIP re-transmit of INVITE for call 524091ef199964d13508e0184586dc1c@24.123.23.170 [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:13383 handle_request_invite: SIP/528-09e02620: This call is UP.... [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:6354 transmit_response_with_sdp: Setting framing from config on incoming call [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:6122 add_sdp: ** Our capability: 0xc (ulaw|alaw) Video flag: True [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:6123 add_sdp: ** Our prefcodec: 0x4 (ulaw) [Feb 22 17:24:21] Audio is at 24.123.23.170 port 16212 [Feb 22 17:24:21] Adding codec 0x4 (ulaw) to SDP [Feb 22 17:24:21] Adding codec 0x8 (alaw) to SDP [Feb 22 17:24:21] Adding non-codec 0x1 (telephone-event) to SDP [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:6254 add_sdp: -- Done with adding codecs to SDP [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:6299 add_sdp: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Feb 22 17:24:21] <--- Reliably 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=z9hG4bKm7ef1a99297826597cbf65730292cc528;received=192.168.1.93 From: 528 ;tag=81c5c0cdcd26df8f2948fc3e1971f0eb To: ;tag=as449238ce Call-ID: 524091ef199964d13508e0184586dc1c@24.123.23.170 CSeq: 440469 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 264 v=0 o=root 14736 14739 IN IP4 192.168.1.66 s=session c=IN IP4 192.168.1.66 t=0 0 m=audio 30006 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 - - - - a=ptime:20 a=sendrecv <------------> [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #207 [Feb 22 17:24:21] <--- 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=z9hG4bKt6beb86d6f9327fe46270f946b85d40ea CSeq: 73555 REGISTER Call-ID: 55ae664767e0@24.123.23.170 From: ;tag=4c2fecd4d2340bbbaadd498c30aca938 To: Contact: ;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="23dc9c01", algorithm=MD5, uri="sip:24.123.23.170", username="523", response="38d5f629f4eb265a0ec8882f3c0f30e5" <-------------> [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKt6beb86d6f9327fe46270f946b85d40ea (82) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: CSeq: 73555 REGISTER (20) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: Call-ID: 55ae664767e0@24.123.23.170 (35) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: ;tag=4c2fecd4d2340bbbaadd498c30aca938 (66) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: To: (27) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="23dc9c01", algorithm=MD5, uri="sip:24.123.23.170", username="523", response="38d5f629f4eb265a0ec8882f3c0f30e5" (167) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 13: (0) [Feb 22 17:24:21] --- (13 headers 0 lines) --- [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:24:21] Using latest REGISTER request as basis request [Feb 22 17:24:21] Sending to 192.168.1.85 : 5060 (no NAT) [Feb 22 17:24:21] <--- 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=z9hG4bKt6beb86d6f9327fe46270f946b85d40ea;received=192.168.1.85 From: ;tag=4c2fecd4d2340bbbaadd498c30aca938 To: Call-ID: 55ae664767e0@24.123.23.170 CSeq: 73555 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:24:21] <--- 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=z9hG4bKt6beb86d6f9327fe46270f946b85d40ea;received=192.168.1.85 From: ;tag=4c2fecd4d2340bbbaadd498c30aca938 To: ;tag=as49f11a59 Call-ID: 55ae664767e0@24.123.23.170 CSeq: 73555 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Thu, 22 Feb 2007 22:24:21 GMT Content-Length: 0 <------------> [Feb 22 17:24:21] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/523 [Feb 22 17:24:21] Scheduling destruction of SIP dialog '55ae664767e0@24.123.23.170' in 32000 ms (Method: REGISTER) [Feb 22 17:24:21] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 523 [Feb 22 17:24:21] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 523 [Feb 22 17:24:21] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/523 - state 1 (Not in use) [Feb 22 17:24:21] DEBUG[14837]: app_queue.c:546 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. [Feb 22 17:24:21] <--- SIP read from 192.168.1.93:5060 ---> ACK sip:526@24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKx25b3a49c8ec0e1300e1f388612957d79 CSeq: 440469 ACK To: ;tag=as449238ce Call-ID: 524091ef199964d13508e0184586dc1c@24.123.23.170 From: 528 ;tag=81c5c0cdcd26df8f2948fc3e1971f0eb User-Agent: Uniden SIP Phone p2 Ver BS4.77 <-------------> [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: ACK sip:526@24.123.23.170 SIP/2.0 (33) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKx25b3a49c8ec0e1300e1f388612957d79 (82) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: CSeq: 440469 ACK (16) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: ;tag=as449238ce (42) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 524091ef199964d13508e0184586dc1c@24.123.23.170 (55) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: From: 528 ;tag=81c5c0cdcd26df8f2948fc3e1971f0eb (74) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: (0) [Feb 22 17:24:21] --- (7 headers 0 lines) --- [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received ACK (6) - Command in SIP ACK [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:2061 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #207 [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '524091ef199964d13508e0184586dc1c@24.123.23.170' of Response 440469: Match Not Found [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:1848 retrans_pkt: SIP TIMER: Rescheduling retransmission #199 (1) INVITE - 5 [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:1862 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #199)) [Feb 22 17:24:21] Retransmitting #1 (NAT) to 192.168.1.91:5060: INVITE sip:550@192.168.1.91:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK7d6ba559;rport From: ;tag=as2b86d5e2 To: "Display Name" ;tag=2772323005 Contact: Call-ID: 1735147470@192.168.1.91 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 266 v=0 o=root 14736 14737 IN IP4 24.123.23.170 s=session c=IN IP4 24.123.23.170 t=0 0 m=audio 15940 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 - - - - a=ptime:20 a=sendrecv --- [Feb 22 17:24:21] DEBUG[14753]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel IAX2/651 [Feb 22 17:24:21] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for IAX2 - 651 [Feb 22 17:24:21] DEBUG[14741]: chan_iax2.c:9600 iax2_devicestate: Checking device state for device 651 [Feb 22 17:24:21] DEBUG[14741]: chan_iax2.c:9608 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 651? addr=-1459509056, defaddr=0 maxms=0, lastms=0 [Feb 22 17:24:21] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for IAX2/651 - state 1 (Not in use) [Feb 22 17:24:21] DEBUG[14838]: app_queue.c:546 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. [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:1848 retrans_pkt: SIP TIMER: Rescheduling retransmission #201 (1) SIP/2.0 - 1 [Feb 22 17:24:21] DEBUG[14767]: chan_sip.c:1862 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #201)) [Feb 22 17:24:21] Retransmitting #1 (no NAT) to 192.168.1.93:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKm7ef1a99297826597cbf65730292cc528;received=192.168.1.93 From: 528 ;tag=81c5c0cdcd26df8f2948fc3e1971f0eb To: ;tag=as449238ce Call-ID: 524091ef199964d13508e0184586dc1c@24.123.23.170 CSeq: 440469 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 264 v=0 o=root 14736 14738 IN IP4 192.168.1.66 s=session c=IN IP4 192.168.1.66 t=0 0 m=audio 30006 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 - - - - a=ptime:20 a=sendrecv --- [Feb 22 17:24:22] DEBUG[14767]: chan_sip.c:1848 retrans_pkt: SIP TIMER: Rescheduling retransmission #199 (2) INVITE - 5 [Feb 22 17:24:22] DEBUG[14767]: chan_sip.c:1862 retrans_pkt: ** SIP timers: Rescheduling retransmission 3 to 2000 ms (t1 500 ms (Retrans id #199)) [Feb 22 17:24:22] Retransmitting #2 (NAT) to 192.168.1.91:5060: INVITE sip:550@192.168.1.91:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK7d6ba559;rport From: ;tag=as2b86d5e2 To: "Display Name" ;tag=2772323005 Contact: Call-ID: 1735147470@192.168.1.91 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 266 v=0 o=root 14736 14737 IN IP4 24.123.23.170 s=session c=IN IP4 24.123.23.170 t=0 0 m=audio 15940 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 - - - - a=ptime:20 a=sendrecv --- [Feb 22 17:24:22] DEBUG[14767]: chan_sip.c:1848 retrans_pkt: SIP TIMER: Rescheduling retransmission #201 (2) SIP/2.0 - 1 [Feb 22 17:24:22] DEBUG[14767]: chan_sip.c:1862 retrans_pkt: ** SIP timers: Rescheduling retransmission 3 to 2000 ms (t1 500 ms (Retrans id #201)) [Feb 22 17:24:22] Retransmitting #2 (no NAT) to 192.168.1.93:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKm7ef1a99297826597cbf65730292cc528;received=192.168.1.93 From: 528 ;tag=81c5c0cdcd26df8f2948fc3e1971f0eb To: ;tag=as449238ce Call-ID: 524091ef199964d13508e0184586dc1c@24.123.23.170 CSeq: 440469 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 264 v=0 o=root 14736 14738 IN IP4 192.168.1.66 s=session c=IN IP4 192.168.1.66 t=0 0 m=audio 30006 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 - - - - a=ptime:20 a=sendrecv --- [Feb 22 17:24:22] DEBUG[14828]: rtp.c:862 ast_rtcp_read: Got RTCP report of 64 bytes [Feb 22 17:24:22] <--- SIP read from 192.168.1.93:5060 ---> ACK sip:526@24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKx25b3a49c8ec0e1300e1f388612957d79 CSeq: 440469 ACK To: ;tag=as449238ce Call-ID: 524091ef199964d13508e0184586dc1c@24.123.23.170 From: 528 ;tag=81c5c0cdcd26df8f2948fc3e1971f0eb User-Agent: Uniden SIP Phone p2 Ver BS4.77 <-------------> [Feb 22 17:24:22] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: ACK sip:526@24.123.23.170 SIP/2.0 (33) [Feb 22 17:24:22] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKx25b3a49c8ec0e1300e1f388612957d79 (82) [Feb 22 17:24:22] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: CSeq: 440469 ACK (16) [Feb 22 17:24:22] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: ;tag=as449238ce (42) [Feb 22 17:24:22] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 524091ef199964d13508e0184586dc1c@24.123.23.170 (55) [Feb 22 17:24:22] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: From: 528 ;tag=81c5c0cdcd26df8f2948fc3e1971f0eb (74) [Feb 22 17:24:22] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:24:22] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: (0) [Feb 22 17:24:22] --- (7 headers 0 lines) --- [Feb 22 17:24:22] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received ACK (6) - Command in SIP ACK [Feb 22 17:24:23] <--- SIP read from 192.168.1.91:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK7d6ba559;rport=5060 From: ;tag=as2b86d5e2 To: "Display Name" ;tag=2772323005 Call-ID: 1735147470@192.168.1.91 CSeq: 102 INVITE Contact: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, MESSAGE, REGISTER, REFER, PING, NOTIFY, INFO, PRACK Content-Type: application/sdp Content-Length: 223 v=0 o=550 123456 654323 IN IP4 192.168.1.91 s=none c=IN IP4 192.168.1.91 t=0 0 m=audio 10010 RTP/AVP 0 8 101 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK7d6ba559;rport=5060 (69) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: ;tag=as2b86d5e2 (49) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: "Display Name" ;tag=2772323005 (62) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 1735147470@192.168.1.91 (32) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: CSeq: 102 INVITE (16) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: (36) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, MESSAGE, REGISTER, REFER, PING, NOTIFY, INFO, PRACK (93) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Content-Type: application/sdp (29) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 223 (21) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: (0) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: v=0 (3) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: o=550 123456 654323 IN IP4 192.168.1.91 (39) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: s=none (6) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: c=IN IP4 192.168.1.91 (21) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: t=0 0 (5) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: m=audio 10010 RTP/AVP 0 8 101 (29) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=ptime:20 (10) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=fmtp:101 0-16 (15) [Feb 22 17:24:23] --- (10 headers 11 lines) --- [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:2053 __sip_ack: Acked pending invite 102 [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:2061 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #199 [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '1735147470@192.168.1.91' of Request 102: Match Not Found [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:11529 handle_response_invite: SIP response 200 to RE-invite on outgoing call 1735147470@192.168.1.91 [Feb 22 17:24:23] Found RTP audio format 0 [Feb 22 17:24:23] Found RTP audio format 8 [Feb 22 17:24:23] Found RTP audio format 101 [Feb 22 17:24:23] Peer audio RTP is at port 192.168.1.91:10010 [Feb 22 17:24:23] Found description format PCMU for ID 0 [Feb 22 17:24:23] Found description format PCMA for ID 8 [Feb 22 17:24:23] Found description format telephone-event for ID 101 [Feb 22 17:24:23] Got unsupported a:fmtp in SDP offer [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:5068 process_sdp: T38 state changed to 0 on channel SIP/550-09e07bc8 [Feb 22 17:24:23] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) [Feb 22 17:24:23] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Feb 22 17:24:23] Peer audio RTP is at port 192.168.1.91:10010 [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:5145 process_sdp: We're settling with these formats: 0xc (ulaw|alaw) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:5152 process_sdp: We have an owner, now see if we need to change this call [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:2978 update_call_counter: Updating call counter for outgoing call [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:7891 build_route: build_route: Contact hop: [Feb 22 17:24:23] list_route: hop: [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:11662 handle_response_invite: Strange... The other side of the bridge does not have a udptl struct [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:11667 handle_response_invite: T38 state changed to 0 on channel SIP [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:11670 handle_response_invite: T38 state changed to 0 on channel SIP/550-09e07bc8 [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:5576 reqprep: Strict routing enforced for session 1735147470@192.168.1.91 [Feb 22 17:24:23] set_destination: Parsing for address/port to send to [Feb 22 17:24:23] set_destination: set destination to 192.168.1.91, port 5060 [Feb 22 17:24:23] Transmitting (NAT) to 192.168.1.91:5060: ACK sip:550@192.168.1.91:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK7abbaaf3;rport From: ;tag=as2b86d5e2 To: "Display Name" ;tag=2772323005 Contact: Call-ID: 1735147470@192.168.1.91 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Feb 22 17:24:23] DEBUG[14828]: rtp.c:862 ast_rtcp_read: Got RTCP report of 32 bytes [Feb 22 17:24:23] DEBUG[14828]: rtp.c:862 ast_rtcp_read: Got RTCP report of 8 bytes [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: OPTIONS sip:sip.broadvoice.com SIP/2.0 (38) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5e2169eb;rport (64) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: "asterisk" ;tag=as2ec11205 (60) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (28) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Contact: (37) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Call-ID: 3a95fe896f9edea13525babd455908ba@24.123.23.170 (55) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Date: Thu, 22 Feb 2007 22:24:23 GMT (35) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Supported: replaces (19) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: Content-Length: 0 (17) [Feb 22 17:24:23] 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=z9hG4bK5e2169eb;rport From: "asterisk" ;tag=as2ec11205 To: Contact: Call-ID: 3a95fe896f9edea13525babd455908ba@24.123.23.170 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 22 Feb 2007 22:24:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #211 [Feb 22 17:24:23] <--- SIP read from 147.135.12.128:5060 ---> SIP/2.0 200 OK Call-ID: 3a95fe896f9edea13525babd455908ba@24.123.23.170 CSeq: 102 OPTIONS From: "asterisk" ;tag=as2ec11205 To: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5e2169eb Supported: 100rel Allow: INVITE, BYE, ACK, OPTIONS, CANCEL, PRACK Accept: application/sdp Accept-Encoding: Accept-Language: en Content-Length: 0 <-------------> [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Call-ID: 3a95fe896f9edea13525babd455908ba@24.123.23.170 (55) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: CSeq: 102 OPTIONS (17) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: From: "asterisk" ;tag=as2ec11205 (60) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: To: (28) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5e2169eb (58) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Supported: 100rel (17) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: Allow: INVITE, BYE, ACK, OPTIONS, CANCEL, PRACK (47) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Accept: application/sdp (23) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Accept-Encoding: (17) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Accept-Language: en (19) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Content-Length: 0 (20) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: (0) [Feb 22 17:24:23] --- (12 headers 0 lines) --- [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:2061 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #211 [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '3a95fe896f9edea13525babd455908ba@24.123.23.170' of Request 102: Match Not Found [Feb 22 17:24:23] Really destroying SIP dialog '3a95fe896f9edea13525babd455908ba@24.123.23.170' Method: OPTIONS [Feb 22 17:24:23] <--- SIP read from 192.168.1.91:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK7d6ba559;rport=5060 From: ;tag=as2b86d5e2 To: "Display Name" ;tag=2772323005 Call-ID: 1735147470@192.168.1.91 CSeq: 102 INVITE Contact: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, MESSAGE, REGISTER, REFER, PING, NOTIFY, INFO, PRACK Content-Type: application/sdp Content-Length: 223 v=0 o=550 123456 654323 IN IP4 192.168.1.91 s=none c=IN IP4 192.168.1.91 t=0 0 m=audio 10010 RTP/AVP 0 8 101 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK7d6ba559;rport=5060 (69) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: ;tag=as2b86d5e2 (49) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: "Display Name" ;tag=2772323005 (62) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 1735147470@192.168.1.91 (32) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: CSeq: 102 INVITE (16) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: (36) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, MESSAGE, REGISTER, REFER, PING, NOTIFY, INFO, PRACK (93) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Content-Type: application/sdp (29) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 223 (21) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: (0) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: v=0 (3) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: o=550 123456 654323 IN IP4 192.168.1.91 (39) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: s=none (6) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: c=IN IP4 192.168.1.91 (21) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: t=0 0 (5) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: m=audio 10010 RTP/AVP 0 8 101 (29) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=ptime:20 (10) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=fmtp:101 0-16 (15) [Feb 22 17:24:23] --- (10 headers 11 lines) --- [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '1735147470@192.168.1.91' of Request 102: Match Found [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:11529 handle_response_invite: SIP response 200 to RE-invite on outgoing call 1735147470@192.168.1.91 [Feb 22 17:24:23] Found RTP audio format 0 [Feb 22 17:24:23] Found RTP audio format 8 [Feb 22 17:24:23] Found RTP audio format 101 [Feb 22 17:24:23] Peer audio RTP is at port 192.168.1.91:10010 [Feb 22 17:24:23] Found description format PCMU for ID 0 [Feb 22 17:24:23] Found description format PCMA for ID 8 [Feb 22 17:24:23] Found description format telephone-event for ID 101 [Feb 22 17:24:23] Got unsupported a:fmtp in SDP offer [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:5068 process_sdp: T38 state changed to 0 on channel SIP/550-09e07bc8 [Feb 22 17:24:23] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) [Feb 22 17:24:23] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Feb 22 17:24:23] Peer audio RTP is at port 192.168.1.91:10010 [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:5145 process_sdp: We're settling with these formats: 0xc (ulaw|alaw) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:5152 process_sdp: We have an owner, now see if we need to change this call [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:2978 update_call_counter: Updating call counter for outgoing call [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:7830 build_route: build_route: Retaining previous route: [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:11662 handle_response_invite: Strange... The other side of the bridge does not have a udptl struct [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:11667 handle_response_invite: T38 state changed to 0 on channel SIP [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:11670 handle_response_invite: T38 state changed to 0 on channel SIP/550-09e07bc8 [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:5576 reqprep: Strict routing enforced for session 1735147470@192.168.1.91 [Feb 22 17:24:23] set_destination: Parsing for address/port to send to [Feb 22 17:24:23] set_destination: set destination to 192.168.1.91, port 5060 [Feb 22 17:24:23] Transmitting (NAT) to 192.168.1.91:5060: ACK sip:550@192.168.1.91:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK6a85e0e8;rport From: ;tag=as2b86d5e2 To: "Display Name" ;tag=2772323005 Contact: Call-ID: 1735147470@192.168.1.91 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Feb 22 17:24:23] <--- SIP read from 192.168.1.91:5060 ---> BYE sip:204@24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.91:5060;rport;branch=z9hG4bK4239596394 From: "Display Name" ;tag=2772323005 To: ;tag=as2b86d5e2 Call-ID: 1735147470@192.168.1.91 CSeq: 1002 BYE max-forwards: 70 user-agent: UTSTARCOM F3000/Device ID-F3000_TEST Content-Length: 0 <-------------> [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: BYE sip:204@24.123.23.170 SIP/2.0 (33) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.91:5060;rport;branch=z9hG4bK4239596394 (65) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: "Display Name" ;tag=2772323005 (64) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: ;tag=as2b86d5e2 (47) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 1735147470@192.168.1.91 (32) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: CSeq: 1002 BYE (14) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: max-forwards: 70 (16) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: user-agent: UTSTARCOM F3000/Device ID-F3000_TEST (48) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Content-Length: 0 (17) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: (0) [Feb 22 17:24:23] --- (9 headers 0 lines) --- [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received BYE (8) - Command in SIP BYE [Feb 22 17:24:23] Sending to 192.168.1.91 : 5060 (NAT) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:1615 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 1735147470@192.168.1.91 [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:14030 handle_request_bye: Received bye, issuing owner hangup .[Feb 22 17:24:23] <--- Transmitting (NAT) to 192.168.1.91:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.91:5060;branch=z9hG4bK4239596394;received=192.168.1.91;rport=5060 From: "Display Name" ;tag=2772323005 To: ;tag=as2b86d5e2 Call-ID: 1735147470@192.168.1.91 CSeq: 1002 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:24:23] DEBUG[14828]: rtp.c:2832 bridge_native_loop: Oooh, got a hangup [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:16704 sip_set_rtp_peer: Sending reinvite on SIP '1ca0c9900034b45264c7ee1a67dc712c@24.123.23.170' - It's audio soon redirected to IP 24.123.23.170 [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:5576 reqprep: Strict routing enforced for session 1ca0c9900034b45264c7ee1a67dc712c@24.123.23.170 [Feb 22 17:24:23] set_destination: Parsing for address/port to send to [Feb 22 17:24:23] set_destination: set destination to 192.168.1.93, port 5060 [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:6122 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: True [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:6123 add_sdp: ** Our prefcodec: 0x4 (ulaw) [Feb 22 17:24:23] Audio is at 24.123.23.170 port 19576 [Feb 22 17:24:23] Adding codec 0x4 (ulaw) to SDP [Feb 22 17:24:23] Adding non-codec 0x1 (telephone-event) to SDP [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:6254 add_sdp: -- Done with adding codecs to SDP [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:6299 add_sdp: Done building SDP. Settling with this capability: 0x4 (ulaw) [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:1603 initialize_initreq: Initializing already initialized SIP dialog 1ca0c9900034b45264c7ee1a67dc712c@24.123.23.170 (presumably reinvite) [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 0: INVITE sip:528@192.168.1.93:5060 SIP/2.0 (40) [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK6fe2f078 (58) [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 2: From: ;tag=as239b687a (44) [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 3: To: 528 ;tag=177014cec8654efa2f4863c390adf2a0 (72) [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 4: Contact: (32) [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 5: Call-ID: 1ca0c9900034b45264c7ee1a67dc712c@24.123.23.170 (55) [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 6: CSeq: 103 INVITE (16) [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 10: Supported: replaces (19) [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 12: Content-Type: application/sdp (29) [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 13: Content-Length: 242 (19) [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:4527 parse_request: Header 14: (0) [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:4559 parse_request: Line: v=0 (3) [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:4559 parse_request: Line: o=root 14736 14738 IN IP4 24.123.23.170 (39) [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:4559 parse_request: Line: s=session (9) [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:4559 parse_request: Line: c=IN IP4 24.123.23.170 (22) [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:4559 parse_request: Line: t=0 0 (5) [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:4559 parse_request: Line: m=audio 19576 RTP/AVP 0 101 (27) [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:4559 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:4559 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:4559 parse_request: Line: a=fmtp:101 0-16 (15) [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:4559 parse_request: Line: a=silenceSupp:off - - - - (25) [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:4559 parse_request: Line: a=ptime:20 (10) [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:4559 parse_request: Line: a=recvonly (10) [Feb 22 17:24:23] Reliably Transmitting (no NAT) to 192.168.1.93:5060: INVITE sip:528@192.168.1.93:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK6fe2f078 From: ;tag=as239b687a To: 528 ;tag=177014cec8654efa2f4863c390adf2a0 Contact: Call-ID: 1ca0c9900034b45264c7ee1a67dc712c@24.123.23.170 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 242 v=0 o=root 14736 14738 IN IP4 24.123.23.170 s=session c=IN IP4 24.123.23.170 t=0 0 m=audio 19576 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly --- [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #214 [Feb 22 17:24:23] DEBUG[14828]: channel.c:3924 ast_channel_bridge: Returning from native bridge, channels: SIP/550-09e07bc8, SIP/528-09e0cbb8 [Feb 22 17:24:23] DEBUG[14828]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/528-09e0cbb8' [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:3278 sip_hangup: Hangup call SIP/528-09e0cbb8, SIP callid 1ca0c9900034b45264c7ee1a67dc712c@24.123.23.170) [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:3286 sip_hangup: update_call_counter(528) - decrement call limit counter on hangup [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:2978 update_call_counter: Updating call counter for outgoing call [Feb 22 17:24:23] Scheduling destruction of SIP dialog '1ca0c9900034b45264c7ee1a67dc712c@24.123.23.170' in 32000 ms (Method: ACK) [Feb 22 17:24:23] DEBUG[14828]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/528-09e0cbb8 [Feb 22 17:24:23] DEBUG[14828]: rtp.c:1465 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Feb 22 17:24:23] DEBUG[14828]: app_dial.c:1643 dial_exec_full: Exiting with DIALSTATUS=ANSWER. [Feb 22 17:24:23] DEBUG[14828]: pbx.c:2363 __ast_pbx_run: Spawn extension (smvoice-sip,204,6) exited non-zero on 'SIP/550-09e07bc8' [Feb 22 17:24:23] == Spawn extension (smvoice-sip, 204, 6) exited non-zero on 'SIP/550-09e07bc8' [Feb 22 17:24:23] DEBUG[14828]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '"550 550" <550>' [Feb 22 17:24:23] DEBUG[14828]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '550' [Feb 22 17:24:23] DEBUG[14828]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '204' [Feb 22 17:24:23] DEBUG[14828]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'smvoice-sip' [Feb 22 17:24:23] DEBUG[14828]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'SIP/550-09e07bc8' [Feb 22 17:24:23] DEBUG[14828]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'SIP/528-09e0cbb8' [Feb 22 17:24:23] DEBUG[14828]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'Dial' [Feb 22 17:24:23] DEBUG[14828]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'SIP/528|20|' [Feb 22 17:24:23] DEBUG[14828]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '2007-02-22 17:24:09' [Feb 22 17:24:23] DEBUG[14828]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '2007-02-22 17:24:13' [Feb 22 17:24:23] DEBUG[14828]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '2007-02-22 17:24:23' [Feb 22 17:24:23] DEBUG[14828]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '14' [Feb 22 17:24:23] DEBUG[14828]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '10' [Feb 22 17:24:23] DEBUG[14828]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'ANSWERED' [Feb 22 17:24:23] DEBUG[14828]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' [Feb 22 17:24:23] DEBUG[14828]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '' [Feb 22 17:24:23] DEBUG[14828]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '1172183049.2' [Feb 22 17:24:23] DEBUG[14828]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '' [Feb 22 17:24:23] DEBUG[14828]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/550-09e07bc8' [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:3278 sip_hangup: Hangup call SIP/550-09e07bc8, SIP callid 1735147470@192.168.1.91) [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:3286 sip_hangup: update_call_counter(550) - decrement call limit counter on hangup [Feb 22 17:24:23] DEBUG[14828]: chan_sip.c:2978 update_call_counter: Updating call counter for outgoing call [Feb 22 17:24:23] DEBUG[14828]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/550-09e07bc8 [Feb 22 17:24:23] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 528 [Feb 22 17:24:23] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 528 [Feb 22 17:24:23] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/528 - state 1 (Not in use) [Feb 22 17:24:23] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 550 [Feb 22 17:24:23] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 550 [Feb 22 17:24:23] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/550 - state 1 (Not in use) [Feb 22 17:24:23] DEBUG[14839]: app_queue.c:546 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. [Feb 22 17:24:23] DEBUG[14840]: app_queue.c:546 changethread: Device 'SIP/550' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Feb 22 17:24:23] <--- SIP read from 192.168.1.93:5060 ---> SIP/2.0 200 OK To: 528 ;tag=177014cec8654efa2f4863c390adf2a0 Contact: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK6fe2f078 From: ;tag=as239b687a Call-ID: 1ca0c9900034b45264c7ee1a67dc712c@24.123.23.170 CSeq: 103 INVITE Content-Length: 242 Supported: sip-cc, sip-cc-01, replaces, timer Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK v=0 o=- 14736 345900 IN IP4 192.168.1.93 s=session c=IN IP4 0.0.0.0 t=0 0 m=audio 30032 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 - - - - a=ptime:20 a=sendonly <-------------> [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: To: 528 ;tag=177014cec8654efa2f4863c390adf2a0 (72) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: Contact: (36) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK6fe2f078 (58) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: ;tag=as239b687a (44) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Call-ID: 1ca0c9900034b45264c7ee1a67dc712c@24.123.23.170 (55) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: CSeq: 103 INVITE (16) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: Content-Length: 242 (19) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: (0) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: v=0 (3) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: o=- 14736 345900 IN IP4 192.168.1.93 (36) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: s=session (9) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: c=IN IP4 0.0.0.0 (16) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: t=0 0 (5) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: m=audio 30032 RTP/AVP 0 101 (27) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=fmtp:101 0-15,16,17,18 (24) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=silenceSupp:off - - - - (25) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=ptime:20 (10) [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=sendonly (10) [Feb 22 17:24:23] --- (10 headers 12 lines) --- [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:2053 __sip_ack: Acked pending invite 103 [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:2061 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #214 [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '1ca0c9900034b45264c7ee1a67dc712c@24.123.23.170' of Request 103: Match Not Found [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:11531 handle_response_invite: SIP response 200 to standard invite [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:2978 update_call_counter: Updating call counter for outgoing call [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:7830 build_route: build_route: Retaining previous route: [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:5576 reqprep: Strict routing enforced for session 1ca0c9900034b45264c7ee1a67dc712c@24.123.23.170 [Feb 22 17:24:23] set_destination: Parsing for address/port to send to [Feb 22 17:24:23] set_destination: set destination to 192.168.1.93, port 5060 [Feb 22 17:24:23] Transmitting (no NAT) to 192.168.1.93:5060: ACK sip:528@192.168.1.93:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK6c537151 From: ;tag=as239b687a To: 528 ;tag=177014cec8654efa2f4863c390adf2a0 Contact: Call-ID: 1ca0c9900034b45264c7ee1a67dc712c@24.123.23.170 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:5576 reqprep: Strict routing enforced for session 1ca0c9900034b45264c7ee1a67dc712c@24.123.23.170 [Feb 22 17:24:23] set_destination: Parsing for address/port to send to [Feb 22 17:24:23] set_destination: set destination to 192.168.1.93, port 5060 [Feb 22 17:24:23] Reliably Transmitting (no NAT) to 192.168.1.93:5060: BYE sip:528@192.168.1.93:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK30736c41 From: ;tag=as239b687a To: 528 ;tag=177014cec8654efa2f4863c390adf2a0 Call-ID: 1ca0c9900034b45264c7ee1a67dc712c@24.123.23.170 CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Feb 22 17:24:23] DEBUG[14767]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #216 [Feb 22 17:24:23] Scheduling destruction of SIP dialog '1ca0c9900034b45264c7ee1a67dc712c@24.123.23.170' in 32000 ms (Method: ACK) [Feb 22 17:24:23] Really destroying SIP dialog '1735147470@192.168.1.91' Method: BYE [Feb 22 17:24:24] <--- SIP read from 192.168.1.93:5060 ---> SIP/2.0 200 OK To: 528 ;tag=177014cec8654efa2f4863c390adf2a0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK30736c41 From: ;tag=as239b687a Call-ID: 1ca0c9900034b45264c7ee1a67dc712c@24.123.23.170 CSeq: 104 BYE Content-Length: 0 Supported: sip-cc, sip-cc-01, replaces, timer Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK <-------------> [Feb 22 17:24:24] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 22 17:24:24] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: To: 528 ;tag=177014cec8654efa2f4863c390adf2a0 (72) [Feb 22 17:24:24] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK30736c41 (58) [Feb 22 17:24:24] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: From: ;tag=as239b687a (44) [Feb 22 17:24:24] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 1ca0c9900034b45264c7ee1a67dc712c@24.123.23.170 (55) [Feb 22 17:24:24] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: CSeq: 104 BYE (13) [Feb 22 17:24:24] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Content-Length: 0 (17) [Feb 22 17:24:24] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:24:24] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:24:24] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: (0) [Feb 22 17:24:24] --- (9 headers 0 lines) --- [Feb 22 17:24:24] DEBUG[14767]: chan_sip.c:2061 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #216 [Feb 22 17:24:24] DEBUG[14767]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '1ca0c9900034b45264c7ee1a67dc712c@24.123.23.170' of Request 104: Match Not Found [Feb 22 17:24:24] Really destroying SIP dialog '1ca0c9900034b45264c7ee1a67dc712c@24.123.23.170' Method: ACK [Feb 22 17:24:24] DEBUG[14767]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '55ae6647666e@24.123.23.170' [Feb 22 17:24:24] DEBUG[14767]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 55ae6647666e@24.123.23.170 [Feb 22 17:24:24] Really destroying SIP dialog '55ae6647666e@24.123.23.170' Method: REGISTER [Feb 22 17:24:24] DEBUG[14767]: chan_sip.c:1848 retrans_pkt: SIP TIMER: Rescheduling retransmission #201 (3) SIP/2.0 - 1 [Feb 22 17:24:24] DEBUG[14767]: chan_sip.c:1862 retrans_pkt: ** SIP timers: Rescheduling retransmission 4 to 4000 ms (t1 500 ms (Retrans id #201)) [Feb 22 17:24:24] Retransmitting #3 (no NAT) to 192.168.1.93:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKm7ef1a99297826597cbf65730292cc528;received=192.168.1.93 From: 528 ;tag=81c5c0cdcd26df8f2948fc3e1971f0eb To: ;tag=as449238ce Call-ID: 524091ef199964d13508e0184586dc1c@24.123.23.170 CSeq: 440469 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 264 v=0 o=root 14736 14738 IN IP4 192.168.1.66 s=session c=IN IP4 192.168.1.66 t=0 0 m=audio 30006 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 - - - - a=ptime:20 a=sendrecv --- [Feb 22 17:24:25] DEBUG[14767]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '2731658924@192.168.1.91' [Feb 22 17:24:25] DEBUG[14767]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 2731658924@192.168.1.91 [Feb 22 17:24:25] Really destroying SIP dialog '2731658924@192.168.1.91' Method: REGISTER [Feb 22 17:24:25] DEBUG[14757]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel IAX2/614 [Feb 22 17:24:25] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for IAX2 - 614 [Feb 22 17:24:25] DEBUG[14741]: chan_iax2.c:9600 iax2_devicestate: Checking device state for device 614 [Feb 22 17:24:25] DEBUG[14741]: chan_iax2.c:9608 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 614? addr=-1493063488, defaddr=0 maxms=0, lastms=0 [Feb 22 17:24:25] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for IAX2/614 - state 1 (Not in use) [Feb 22 17:24:25] DEBUG[14841]: app_queue.c:546 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. [Feb 22 17:24:26] <--- 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=z9hG4bKfbb451dd51002b1c974543250cc05dc92 Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 26672 REGISTER From: ;tag=3f375eb8a461f2d96e9ca9c6d69612a3 To: Contact: ;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="4ee04135", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="6df9ba88fbf3ecee38524e2f12d5dace" <-------------> [Feb 22 17:24:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:24:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKfbb451dd51002b1c974543250cc05dc92 (82) [Feb 22 17:24:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: Call-ID: 55ae66469f7f@24.123.23.170 (35) [Feb 22 17:24:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: CSeq: 26672 REGISTER (20) [Feb 22 17:24:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: ;tag=3f375eb8a461f2d96e9ca9c6d69612a3 (66) [Feb 22 17:24:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: To: (27) [Feb 22 17:24:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Feb 22 17:24:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:24:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 22 17:24:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:24:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:24:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:24:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="4ee04135", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="6df9ba88fbf3ecee38524e2f12d5dace" (167) [Feb 22 17:24:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 13: (0) [Feb 22 17:24:26] --- (13 headers 0 lines) --- [Feb 22 17:24:26] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 55ae66469f7f@24.123.23.170 - REGISTER (No RTP) [Feb 22 17:24:26] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:24:26] Using latest REGISTER request as basis request [Feb 22 17:24:26] Sending to 192.168.1.93 : 5060 (no NAT) [Feb 22 17:24:26] <--- 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=z9hG4bKfbb451dd51002b1c974543250cc05dc92;received=192.168.1.93 From: ;tag=3f375eb8a461f2d96e9ca9c6d69612a3 To: Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 26672 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:24:26] <--- 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=z9hG4bKfbb451dd51002b1c974543250cc05dc92;received=192.168.1.93 From: ;tag=3f375eb8a461f2d96e9ca9c6d69612a3 To: ;tag=as0206583d Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 26672 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6610b22c" Content-Length: 0 <------------> [Feb 22 17:24:26] Scheduling destruction of SIP dialog '55ae66469f7f@24.123.23.170' in 32000 ms (Method: REGISTER) [Feb 22 17:24:26] <--- SIP read from 192.168.1.86:5060 ---> <-------------> [Feb 22 17:24:26] --- (0 headers 0 lines) Nat keepalive --- [Feb 22 17:24:26] <--- 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=z9hG4bKje3084779a9b9acae3e423a4cf266c484 CSeq: 26673 REGISTER Call-ID: 55ae66469f7f@24.123.23.170 From: ;tag=3f375eb8a461f2d96e9ca9c6d69612a3 To: Contact: ;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="6610b22c", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="ba10fdfe1cf81a15ad4b66ddfd2bbb9e" <-------------> [Feb 22 17:24:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:24:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKje3084779a9b9acae3e423a4cf266c484 (82) [Feb 22 17:24:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: CSeq: 26673 REGISTER (20) [Feb 22 17:24:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: Call-ID: 55ae66469f7f@24.123.23.170 (35) [Feb 22 17:24:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: ;tag=3f375eb8a461f2d96e9ca9c6d69612a3 (66) [Feb 22 17:24:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: To: (27) [Feb 22 17:24:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Feb 22 17:24:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:24:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 22 17:24:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:24:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:24:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:24:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="6610b22c", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="ba10fdfe1cf81a15ad4b66ddfd2bbb9e" (167) [Feb 22 17:24:26] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 13: (0) [Feb 22 17:24:26] --- (13 headers 0 lines) --- [Feb 22 17:24:26] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:24:26] Using latest REGISTER request as basis request [Feb 22 17:24:26] Sending to 192.168.1.93 : 5060 (no NAT) [Feb 22 17:24:26] <--- 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=z9hG4bKje3084779a9b9acae3e423a4cf266c484;received=192.168.1.93 From: ;tag=3f375eb8a461f2d96e9ca9c6d69612a3 To: Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 26673 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:24:26] <--- 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=z9hG4bKje3084779a9b9acae3e423a4cf266c484;received=192.168.1.93 From: ;tag=3f375eb8a461f2d96e9ca9c6d69612a3 To: ;tag=as0206583d Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 26673 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Thu, 22 Feb 2007 22:24:26 GMT Content-Length: 0 <------------> [Feb 22 17:24:26] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/528 [Feb 22 17:24:26] Scheduling destruction of SIP dialog '55ae66469f7f@24.123.23.170' in 32000 ms (Method: REGISTER) [Feb 22 17:24:26] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 528 [Feb 22 17:24:26] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 528 [Feb 22 17:24:26] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/528 - state 1 (Not in use) [Feb 22 17:24:26] DEBUG[14842]: app_queue.c:546 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. [Feb 22 17:24:26] DEBUG[14767]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '000ff78d-ebb20007-74946e6d-3266b768@192.168.1.165' [Feb 22 17:24:26] DEBUG[14767]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 000ff78d-ebb20007-74946e6d-3266b768@192.168.1.165 [Feb 22 17:24:26] Really destroying SIP dialog '000ff78d-ebb20007-74946e6d-3266b768@192.168.1.165' Method: REGISTER [Feb 22 17:24:27] DEBUG[14767]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '55ae664696c0@24.123.23.170' [Feb 22 17:24:27] DEBUG[14767]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 55ae664696c0@24.123.23.170 [Feb 22 17:24:27] Really destroying SIP dialog '55ae664696c0@24.123.23.170' Method: REGISTER [Feb 22 17:24:27] DEBUG[14767]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '42fcb9a2205a226e1657915a20f1c05a@24.123.23.170' [Feb 22 17:24:27] DEBUG[14767]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 42fcb9a2205a226e1657915a20f1c05a@24.123.23.170 [Feb 22 17:24:27] Really destroying SIP dialog '42fcb9a2205a226e1657915a20f1c05a@24.123.23.170' Method: REGISTER [Feb 22 17:24:28] <--- 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=z9hG4bKv119192d19ec64d10fc321b6584088230 Call-ID: 55ae66476793@24.123.23.170 CSeq: 108464 REGISTER From: ;tag=b6b1e12ee819b5b6b1eb466cf906069c To: Contact: ;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="101b6db8", algorithm=MD5, uri="sip:24.123.23.170", username="525", response="fbdcf2731f8b245232f034935deddd98" <-------------> [Feb 22 17:24:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:24:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bKv119192d19ec64d10fc321b6584088230 (82) [Feb 22 17:24:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: Call-ID: 55ae66476793@24.123.23.170 (35) [Feb 22 17:24:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: CSeq: 108464 REGISTER (21) [Feb 22 17:24:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: ;tag=b6b1e12ee819b5b6b1eb466cf906069c (66) [Feb 22 17:24:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: To: (27) [Feb 22 17:24:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Feb 22 17:24:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:24:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 22 17:24:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:24:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:24:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:24:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="101b6db8", algorithm=MD5, uri="sip:24.123.23.170", username="525", response="fbdcf2731f8b245232f034935deddd98" (167) [Feb 22 17:24:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 13: (0) [Feb 22 17:24:28] --- (13 headers 0 lines) --- [Feb 22 17:24:28] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 55ae66476793@24.123.23.170 - REGISTER (No RTP) [Feb 22 17:24:28] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:24:28] Using latest REGISTER request as basis request [Feb 22 17:24:28] Sending to 192.168.1.76 : 5060 (no NAT) [Feb 22 17:24:28] <--- 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=z9hG4bKv119192d19ec64d10fc321b6584088230;received=192.168.1.76 From: ;tag=b6b1e12ee819b5b6b1eb466cf906069c To: Call-ID: 55ae66476793@24.123.23.170 CSeq: 108464 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:24:28] <--- 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=z9hG4bKv119192d19ec64d10fc321b6584088230;received=192.168.1.76 From: ;tag=b6b1e12ee819b5b6b1eb466cf906069c To: ;tag=as212a67ef Call-ID: 55ae66476793@24.123.23.170 CSeq: 108464 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2637effa" Content-Length: 0 <------------> [Feb 22 17:24:28] Scheduling destruction of SIP dialog '55ae66476793@24.123.23.170' in 32000 ms (Method: REGISTER) [Feb 22 17:24:28] <--- SIP read from 68.58.36.157:5060 ---> <-------------> [Feb 22 17:24:28] --- (0 headers 0 lines) Nat keepalive --- [Feb 22 17:24:28] <--- 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=z9hG4bKt192f711912d38831b385ef88bc0e18e9 CSeq: 108465 REGISTER Call-ID: 55ae66476793@24.123.23.170 From: ;tag=b6b1e12ee819b5b6b1eb466cf906069c To: Contact: ;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="2637effa", algorithm=MD5, uri="sip:24.123.23.170", username="525", response="9fc04abeaf29ee5c04616acd20f9c6bc" <-------------> [Feb 22 17:24:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:24:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bKt192f711912d38831b385ef88bc0e18e9 (82) [Feb 22 17:24:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: CSeq: 108465 REGISTER (21) [Feb 22 17:24:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: Call-ID: 55ae66476793@24.123.23.170 (35) [Feb 22 17:24:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: ;tag=b6b1e12ee819b5b6b1eb466cf906069c (66) [Feb 22 17:24:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: To: (27) [Feb 22 17:24:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Feb 22 17:24:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:24:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 22 17:24:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:24:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:24:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:24:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="2637effa", algorithm=MD5, uri="sip:24.123.23.170", username="525", response="9fc04abeaf29ee5c04616acd20f9c6bc" (167) [Feb 22 17:24:28] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 13: (0) [Feb 22 17:24:28] --- (13 headers 0 lines) --- [Feb 22 17:24:28] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:24:28] Using latest REGISTER request as basis request [Feb 22 17:24:28] Sending to 192.168.1.76 : 5060 (no NAT) [Feb 22 17:24:28] <--- 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=z9hG4bKt192f711912d38831b385ef88bc0e18e9;received=192.168.1.76 From: ;tag=b6b1e12ee819b5b6b1eb466cf906069c To: Call-ID: 55ae66476793@24.123.23.170 CSeq: 108465 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:24:28] <--- 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=z9hG4bKt192f711912d38831b385ef88bc0e18e9;received=192.168.1.76 From: ;tag=b6b1e12ee819b5b6b1eb466cf906069c To: ;tag=as212a67ef Call-ID: 55ae66476793@24.123.23.170 CSeq: 108465 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Thu, 22 Feb 2007 22:24:28 GMT Content-Length: 0 <------------> [Feb 22 17:24:28] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/525 [Feb 22 17:24:28] Scheduling destruction of SIP dialog '55ae66476793@24.123.23.170' in 32000 ms (Method: REGISTER) [Feb 22 17:24:28] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 525 [Feb 22 17:24:28] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 525 [Feb 22 17:24:28] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/525 - state 1 (Not in use) [Feb 22 17:24:28] DEBUG[14843]: app_queue.c:546 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. [Feb 22 17:24:28] DEBUG[14767]: chan_sip.c:1848 retrans_pkt: SIP TIMER: Rescheduling retransmission #201 (4) SIP/2.0 - 1 [Feb 22 17:24:28] DEBUG[14767]: chan_sip.c:1862 retrans_pkt: ** SIP timers: Rescheduling retransmission 5 to 4000 ms (t1 500 ms (Retrans id #201)) [Feb 22 17:24:28] Retransmitting #4 (no NAT) to 192.168.1.93:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKm7ef1a99297826597cbf65730292cc528;received=192.168.1.93 From: 528 ;tag=81c5c0cdcd26df8f2948fc3e1971f0eb To: ;tag=as449238ce Call-ID: 524091ef199964d13508e0184586dc1c@24.123.23.170 CSeq: 440469 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 264 v=0 o=root 14736 14738 IN IP4 192.168.1.66 s=session c=IN IP4 192.168.1.66 t=0 0 m=audio 30006 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 - - - - a=ptime:20 a=sendrecv --- [Feb 22 17:24:31] DEBUG[14767]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '000ff78d-ebb20003-42dc795d-3e449dd6@192.168.1.165' [Feb 22 17:24:31] DEBUG[14767]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 000ff78d-ebb20003-42dc795d-3e449dd6@192.168.1.165 [Feb 22 17:24:31] Really destroying SIP dialog '000ff78d-ebb20003-42dc795d-3e449dd6@192.168.1.165' Method: REGISTER [Feb 22 17:24:32] DEBUG[14767]: chan_sip.c:1848 retrans_pkt: SIP TIMER: Rescheduling retransmission #201 (5) SIP/2.0 - 1 [Feb 22 17:24:32] DEBUG[14767]: chan_sip.c:1862 retrans_pkt: ** SIP timers: Rescheduling retransmission 6 to 4000 ms (t1 500 ms (Retrans id #201)) [Feb 22 17:24:32] Retransmitting #5 (no NAT) to 192.168.1.93:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKm7ef1a99297826597cbf65730292cc528;received=192.168.1.93 From: 528 ;tag=81c5c0cdcd26df8f2948fc3e1971f0eb To: ;tag=as449238ce Call-ID: 524091ef199964d13508e0184586dc1c@24.123.23.170 CSeq: 440469 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 264 v=0 o=root 14736 14738 IN IP4 192.168.1.66 s=session c=IN IP4 192.168.1.66 t=0 0 m=audio 30006 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 - - - - a=ptime:20 a=sendrecv --- [Feb 22 17:24:35] DEBUG[14760]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel IAX2/606 [Feb 22 17:24:35] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for IAX2 - 606 [Feb 22 17:24:35] DEBUG[14741]: chan_iax2.c:9600 iax2_devicestate: Checking device state for device 606 [Feb 22 17:24:35] DEBUG[14741]: chan_iax2.c:9608 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 606? addr=-1509840704, defaddr=0 maxms=0, lastms=0 [Feb 22 17:24:35] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for IAX2/606 - state 1 (Not in use) [Feb 22 17:24:35] DEBUG[14844]: app_queue.c:546 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. [Feb 22 17:24:35] DEBUG[14767]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '000ff78d-ebb20005-4fe621ee-067da175@192.168.1.165' [Feb 22 17:24:35] DEBUG[14767]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 000ff78d-ebb20005-4fe621ee-067da175@192.168.1.165 [Feb 22 17:24:35] Really destroying SIP dialog '000ff78d-ebb20005-4fe621ee-067da175@192.168.1.165' Method: REGISTER [Feb 22 17:24:36] DEBUG[14845]: manager.c:1931 process_message: Manager received command 'Login' [Feb 22 17:24:36] == Parsing '/etc/asterisk/manager.conf': [Feb 22 17:24:36] DEBUG[14845]: config.c:844 config_text_file_load: Parsing /etc/asterisk/manager.conf [Feb 22 17:24:36] Found [Feb 22 17:24:36] DEBUG[14845]: acl.c:199 ast_append_ha: 127.0.0.1/255.255.255.0/255.255.255.0 appended to acl for peer [Feb 22 17:24:36] DEBUG[14845]: acl.c:213 ast_apply_ha: ##### Testing 127.0.0.1 with 127.0.0.0 [Feb 22 17:24:36] == Manager 'MessageNet' logged on from 127.0.0.1 [Feb 22 17:24:36] DEBUG[14845]: manager.c:1931 process_message: Manager received command 'Command' [Feb 22 17:24:36] DEBUG[14845]: manager.c:1931 process_message: Manager received command 'Command' [Feb 22 17:24:36] DEBUG[14845]: manager.c:1931 process_message: Manager received command 'Logoff' [Feb 22 17:24:36] == Manager 'MessageNet' logged off from 127.0.0.1 [Feb 22 17:24:36] <--- SIP read from 192.168.1.86:5062 ---> <-------------> [Feb 22 17:24:36] --- (0 headers 0 lines) Nat keepalive --- [Feb 22 17:24:36] DEBUG[14767]: chan_sip.c:1848 retrans_pkt: SIP TIMER: Rescheduling retransmission #201 (6) SIP/2.0 - 1 [Feb 22 17:24:36] DEBUG[14767]: chan_sip.c:1862 retrans_pkt: ** SIP timers: Rescheduling retransmission 7 to 4000 ms (t1 500 ms (Retrans id #201)) [Feb 22 17:24:36] Retransmitting #6 (no NAT) to 192.168.1.93:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKm7ef1a99297826597cbf65730292cc528;received=192.168.1.93 From: 528 ;tag=81c5c0cdcd26df8f2948fc3e1971f0eb To: ;tag=as449238ce Call-ID: 524091ef199964d13508e0184586dc1c@24.123.23.170 CSeq: 440469 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 264 v=0 o=root 14736 14738 IN IP4 192.168.1.66 s=session c=IN IP4 192.168.1.66 t=0 0 m=audio 30006 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 - - - - a=ptime:20 a=sendrecv --- [Feb 22 17:24:40] <--- 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=z9hG4bKxe1e2b34a7cdc344dae21654e230ada4f Call-ID: 55ae66469f11@24.123.23.170 CSeq: 33546 REGISTER From: ;tag=62c064a2c13deb1fc8016a356298ba0d To: Contact: ;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="57cac9ba", algorithm=MD5, uri="sip:24.123.23.170", username="522", response="3e037aefd24e5ff3815a099fac8aa29b" <-------------> [Feb 22 17:24:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:24:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.95:5060;branch=z9hG4bKxe1e2b34a7cdc344dae21654e230ada4f (82) [Feb 22 17:24:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: Call-ID: 55ae66469f11@24.123.23.170 (35) [Feb 22 17:24:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: CSeq: 33546 REGISTER (20) [Feb 22 17:24:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: ;tag=62c064a2c13deb1fc8016a356298ba0d (66) [Feb 22 17:24:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: To: (27) [Feb 22 17:24:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Feb 22 17:24:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:24:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 22 17:24:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:24:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:24:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:24:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="57cac9ba", algorithm=MD5, uri="sip:24.123.23.170", username="522", response="3e037aefd24e5ff3815a099fac8aa29b" (167) [Feb 22 17:24:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 13: (0) [Feb 22 17:24:40] --- (13 headers 0 lines) --- [Feb 22 17:24:40] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 55ae66469f11@24.123.23.170 - REGISTER (No RTP) [Feb 22 17:24:40] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:24:40] Using latest REGISTER request as basis request [Feb 22 17:24:40] Sending to 192.168.1.95 : 5060 (no NAT) [Feb 22 17:24:40] <--- 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=z9hG4bKxe1e2b34a7cdc344dae21654e230ada4f;received=192.168.1.95 From: ;tag=62c064a2c13deb1fc8016a356298ba0d To: Call-ID: 55ae66469f11@24.123.23.170 CSeq: 33546 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:24:40] <--- 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=z9hG4bKxe1e2b34a7cdc344dae21654e230ada4f;received=192.168.1.95 From: ;tag=62c064a2c13deb1fc8016a356298ba0d To: ;tag=as49162067 Call-ID: 55ae66469f11@24.123.23.170 CSeq: 33546 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="749cf649" Content-Length: 0 <------------> [Feb 22 17:24:40] Scheduling destruction of SIP dialog '55ae66469f11@24.123.23.170' in 32000 ms (Method: REGISTER) [Feb 22 17:24:40] DEBUG[14764]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel IAX2/616 [Feb 22 17:24:40] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for IAX2 - 616 [Feb 22 17:24:40] DEBUG[14741]: chan_iax2.c:9600 iax2_devicestate: Checking device state for device 616 [Feb 22 17:24:40] DEBUG[14741]: chan_iax2.c:9608 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 616? addr=-1891793590, defaddr=0 maxms=0, lastms=0 [Feb 22 17:24:40] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for IAX2/616 - state 1 (Not in use) [Feb 22 17:24:40] DEBUG[14846]: app_queue.c:546 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. [Feb 22 17:24:40] <--- 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=z9hG4bKn8f050495ea766f66c1290907b114e0c4 CSeq: 33547 REGISTER Call-ID: 55ae66469f11@24.123.23.170 From: ;tag=62c064a2c13deb1fc8016a356298ba0d To: Contact: ;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="749cf649", algorithm=MD5, uri="sip:24.123.23.170", username="522", response="1224296765e4f152cbd526e202d32611" <-------------> [Feb 22 17:24:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:24:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.95:5060;branch=z9hG4bKn8f050495ea766f66c1290907b114e0c4 (82) [Feb 22 17:24:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: CSeq: 33547 REGISTER (20) [Feb 22 17:24:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: Call-ID: 55ae66469f11@24.123.23.170 (35) [Feb 22 17:24:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: ;tag=62c064a2c13deb1fc8016a356298ba0d (66) [Feb 22 17:24:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: To: (27) [Feb 22 17:24:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Feb 22 17:24:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:24:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 22 17:24:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:24:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:24:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:24:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="749cf649", algorithm=MD5, uri="sip:24.123.23.170", username="522", response="1224296765e4f152cbd526e202d32611" (167) [Feb 22 17:24:40] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 13: (0) [Feb 22 17:24:40] --- (13 headers 0 lines) --- [Feb 22 17:24:40] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:24:40] Using latest REGISTER request as basis request [Feb 22 17:24:40] Sending to 192.168.1.95 : 5060 (no NAT) [Feb 22 17:24:40] <--- 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=z9hG4bKn8f050495ea766f66c1290907b114e0c4;received=192.168.1.95 From: ;tag=62c064a2c13deb1fc8016a356298ba0d To: Call-ID: 55ae66469f11@24.123.23.170 CSeq: 33547 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:24:40] <--- 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=z9hG4bKn8f050495ea766f66c1290907b114e0c4;received=192.168.1.95 From: ;tag=62c064a2c13deb1fc8016a356298ba0d To: ;tag=as49162067 Call-ID: 55ae66469f11@24.123.23.170 CSeq: 33547 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Thu, 22 Feb 2007 22:24:40 GMT Content-Length: 0 <------------> [Feb 22 17:24:40] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/522 [Feb 22 17:24:40] Scheduling destruction of SIP dialog '55ae66469f11@24.123.23.170' in 32000 ms (Method: REGISTER) [Feb 22 17:24:40] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 522 [Feb 22 17:24:40] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 522 [Feb 22 17:24:40] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/522 - state 1 (Not in use) [Feb 22 17:24:40] DEBUG[14847]: app_queue.c:546 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. [Feb 22 17:24:40] WARNING[14767]: chan_sip.c:1881 retrans_pkt: Maximum retries exceeded on transmission 524091ef199964d13508e0184586dc1c@24.123.23.170 for seqno 440469 (Critical Response) [Feb 22 17:24:40] DEBUG[14767]: chan_sip.c:1615 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 524091ef199964d13508e0184586dc1c@24.123.23.170 [Feb 22 17:24:40] WARNING[14767]: chan_sip.c:1898 retrans_pkt: Hanging up call 524091ef199964d13508e0184586dc1c@24.123.23.170 - no reply to our critical packet. [Feb 22 17:24:40] DEBUG[14821]: rtp.c:2832 bridge_native_loop: Oooh, got a hangup [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:16704 sip_set_rtp_peer: Sending reinvite on SIP '903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170' - It's audio soon redirected to IP 24.123.23.170 [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:5576 reqprep: Strict routing enforced for session 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 [Feb 22 17:24:40] set_destination: Parsing for address/port to send to [Feb 22 17:24:40] set_destination: set destination to 192.168.1.66, port 5060 [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:6122 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: True [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:6123 add_sdp: ** Our prefcodec: 0x0 (nothing) [Feb 22 17:24:40] Audio is at 24.123.23.170 port 15904 [Feb 22 17:24:40] Adding codec 0x4 (ulaw) to SDP [Feb 22 17:24:40] Adding non-codec 0x1 (telephone-event) to SDP [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:6254 add_sdp: -- Done with adding codecs to SDP [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:6299 add_sdp: Done building SDP. Settling with this capability: 0x4 (ulaw) [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:1603 initialize_initreq: Initializing already initialized SIP dialog 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 (presumably reinvite) [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 0: INVITE sip:526@192.168.1.66:5060 SIP/2.0 (40) [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5cf29e92 (58) [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 2: From: ;tag=as056fdf38 (44) [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 3: To: 526 ;tag=48106fce45713fff4b942431e90afa0e (68) [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 4: Contact: (32) [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 5: Call-ID: 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 (68) [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 6: CSeq: 104 INVITE (16) [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 10: Supported: replaces (19) [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 12: Content-Type: application/sdp (29) [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 13: Content-Length: 242 (19) [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:4527 parse_request: Header 14: (0) [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: v=0 (3) [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: o=root 14736 14739 IN IP4 24.123.23.170 (39) [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: s=session (9) [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: c=IN IP4 24.123.23.170 (22) [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: t=0 0 (5) [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: m=audio 15904 RTP/AVP 0 101 (27) [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: a=fmtp:101 0-16 (15) [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: a=silenceSupp:off - - - - (25) [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: a=ptime:20 (10) [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:4559 parse_request: Line: a=sendrecv (10) [Feb 22 17:24:40] Reliably Transmitting (no NAT) to 192.168.1.66:5060: INVITE sip:526@192.168.1.66:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5cf29e92 From: ;tag=as056fdf38 To: 526 ;tag=48106fce45713fff4b942431e90afa0e Contact: Call-ID: 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 CSeq: 104 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 242 v=0 o=root 14736 14739 IN IP4 24.123.23.170 s=session c=IN IP4 24.123.23.170 t=0 0 m=audio 15904 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #227 [Feb 22 17:24:40] DEBUG[14821]: channel.c:3924 ast_channel_bridge: Returning from native bridge, channels: SIP/526-09df30e0, SIP/528-09e02620 [Feb 22 17:24:40] DEBUG[14821]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/528-09e02620' [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:3278 sip_hangup: Hangup call SIP/528-09e02620, SIP callid 524091ef199964d13508e0184586dc1c@24.123.23.170) [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:3286 sip_hangup: update_call_counter(528) - decrement call limit counter on hangup [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:2978 update_call_counter: Updating call counter for incoming call [Feb 22 17:24:40] DEBUG[14821]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/528-09e02620 [Feb 22 17:24:40] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 528 [Feb 22 17:24:40] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 528 [Feb 22 17:24:40] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/528 - state 1 (Not in use) [Feb 22 17:24:40] DEBUG[14848]: app_queue.c:546 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. [Feb 22 17:24:40] DEBUG[14821]: rtp.c:1465 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Feb 22 17:24:40] DEBUG[14821]: app_dial.c:1643 dial_exec_full: Exiting with DIALSTATUS=ANSWER. [Feb 22 17:24:40] DEBUG[14821]: pbx.c:2363 __ast_pbx_run: Spawn extension (smvoice-sip,204,6) exited non-zero on 'SIP/526-09df30e0' [Feb 22 17:24:40] == Spawn extension (smvoice-sip, 204, 6) exited non-zero on 'SIP/526-09df30e0' [Feb 22 17:24:40] DEBUG[14821]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '"526 526" <526>' [Feb 22 17:24:40] DEBUG[14821]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '526' [Feb 22 17:24:40] DEBUG[14821]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '204' [Feb 22 17:24:40] DEBUG[14821]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'smvoice-sip' [Feb 22 17:24:40] DEBUG[14821]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'SIP/526-09df30e0' [Feb 22 17:24:40] DEBUG[14821]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'SIP/528-09e02620' [Feb 22 17:24:40] DEBUG[14821]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'Dial' [Feb 22 17:24:40] DEBUG[14821]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'SIP/528|20|' [Feb 22 17:24:40] DEBUG[14821]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '2007-02-22 17:24:00' [Feb 22 17:24:40] DEBUG[14821]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '2007-02-22 17:24:03' [Feb 22 17:24:40] DEBUG[14821]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '2007-02-22 17:24:40' [Feb 22 17:24:40] DEBUG[14821]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '40' [Feb 22 17:24:40] DEBUG[14821]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '37' [Feb 22 17:24:40] DEBUG[14821]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'ANSWERED' [Feb 22 17:24:40] DEBUG[14821]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' [Feb 22 17:24:40] DEBUG[14821]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '' [Feb 22 17:24:40] DEBUG[14821]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '1172183040.0' [Feb 22 17:24:40] DEBUG[14821]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '' [Feb 22 17:24:40] DEBUG[14821]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/526-09df30e0' [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:3278 sip_hangup: Hangup call SIP/526-09df30e0, SIP callid 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170) [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:3286 sip_hangup: update_call_counter(526) - decrement call limit counter on hangup [Feb 22 17:24:40] DEBUG[14821]: chan_sip.c:2978 update_call_counter: Updating call counter for outgoing call [Feb 22 17:24:40] Scheduling destruction of SIP dialog '903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170' in 32000 ms (Method: ACK) [Feb 22 17:24:40] DEBUG[14821]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/526-09df30e0 [Feb 22 17:24:40] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 526 [Feb 22 17:24:40] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 526 [Feb 22 17:24:40] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/526 - state 1 (Not in use) [Feb 22 17:24:40] DEBUG[14849]: app_queue.c:546 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. [Feb 22 17:24:40] Really destroying SIP dialog '524091ef199964d13508e0184586dc1c@24.123.23.170' Method: ACK [Feb 22 17:24:41] <--- SIP read from 192.168.1.66:5060 ---> SIP/2.0 200 OK To: 526 ;tag=48106fce45713fff4b942431e90afa0e Contact: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5cf29e92 From: ;tag=as056fdf38 Call-ID: 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 CSeq: 104 INVITE Content-Type: application/sdp Content-Length: 247 Supported: sip-cc, sip-cc-01, replaces, timer Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK v=0 o=- 14736 163218 IN IP4 192.168.1.66 s=session c=IN IP4 192.168.1.66 t=0 0 m=audio 30006 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 - - - - a=ptime:20 a=sendrecv <-------------> [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: To: 526 ;tag=48106fce45713fff4b942431e90afa0e (68) [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: Contact: (36) [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5cf29e92 (58) [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: ;tag=as056fdf38 (44) [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Call-ID: 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 (68) [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: CSeq: 104 INVITE (16) [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: Content-Type: application/sdp (29) [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Content-Length: 247 (19) [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: (0) [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: v=0 (3) [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: o=- 14736 163218 IN IP4 192.168.1.66 (36) [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: s=session (9) [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: c=IN IP4 192.168.1.66 (21) [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: t=0 0 (5) [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: m=audio 30006 RTP/AVP 0 101 (27) [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=fmtp:101 0-15,16,17,18 (24) [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=silenceSupp:off - - - - (25) [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=ptime:20 (10) [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:4559 parse_request: Line: a=sendrecv (10) [Feb 22 17:24:41] --- (11 headers 12 lines) --- [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:2053 __sip_ack: Acked pending invite 104 [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:2061 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #227 [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170' of Request 104: Match Not Found [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:11531 handle_response_invite: SIP response 200 to standard invite [Feb 22 17:24:41] Found RTP audio format 0 [Feb 22 17:24:41] Found RTP audio format 101 [Feb 22 17:24:41] Peer audio RTP is at port 192.168.1.66:30006 [Feb 22 17:24:41] Found description format PCMU for ID 0 [Feb 22 17:24:41] Found description format telephone-event for ID 101 [Feb 22 17:24:41] Got unsupported a:fmtp in SDP offer [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:5068 process_sdp: T38 state changed to 0 on channel [Feb 22 17:24:41] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Feb 22 17:24:41] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Feb 22 17:24:41] Peer audio RTP is at port 192.168.1.66:30006 [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:5145 process_sdp: We're settling with these formats: 0x4 (ulaw) [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:2978 update_call_counter: Updating call counter for outgoing call [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:7830 build_route: build_route: Retaining previous route: [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:5576 reqprep: Strict routing enforced for session 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 [Feb 22 17:24:41] set_destination: Parsing for address/port to send to [Feb 22 17:24:41] set_destination: set destination to 192.168.1.66, port 5060 [Feb 22 17:24:41] Transmitting (no NAT) to 192.168.1.66:5060: ACK sip:526@192.168.1.66:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK484fe83b From: ;tag=as056fdf38 To: 526 ;tag=48106fce45713fff4b942431e90afa0e Contact: Call-ID: 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 CSeq: 104 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:5576 reqprep: Strict routing enforced for session 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 [Feb 22 17:24:41] set_destination: Parsing for address/port to send to [Feb 22 17:24:41] set_destination: set destination to 192.168.1.66, port 5060 [Feb 22 17:24:41] Reliably Transmitting (no NAT) to 192.168.1.66:5060: BYE sip:526@192.168.1.66:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK0de5499a From: ;tag=as056fdf38 To: 526 ;tag=48106fce45713fff4b942431e90afa0e Call-ID: 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 CSeq: 105 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #229 [Feb 22 17:24:41] Scheduling destruction of SIP dialog '903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170' in 32000 ms (Method: ACK) [Feb 22 17:24:41] <--- SIP read from 192.168.1.66:5060 ---> SIP/2.0 200 OK To: 526 ;tag=48106fce45713fff4b942431e90afa0e Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK0de5499a From: ;tag=as056fdf38 Call-ID: 903b6b4930ad846bcc11b51d65857326-55ae6647666e@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 <-------------> [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: To: 526 ;tag=48106fce45713fff4b942431e90afa0e (68) [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK0de5499a (58) [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: From: ;tag=as056fdf38 (44) [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170 (68) [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: CSeq: 105 BYE (13) [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Content-Length: 0 (17) [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: (0) [Feb 22 17:24:41] --- (9 headers 0 lines) --- [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:2061 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #229 [Feb 22 17:24:41] DEBUG[14767]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170' of Request 105: Match Not Found [Feb 22 17:24:41] Really destroying SIP dialog '903b6b4930ad846bcc11b51d65857326-55ae6647666e@24.123.23.170' Method: ACK [Feb 22 17:24:43] NOTICE[14767]: chan_sip.c:7055 sip_reregister: -- Re-registration for 3173241052@sip.broadvoice.com [Feb 22 17:24:43] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 42fcb9a2205a226e1657915a20f1c05a@24.123.23.170 - REGISTER (No RTP) [Feb 22 17:24:43] DEBUG[14767]: chan_sip.c:7214 transmit_register: Scheduled a registration timeout for sip.broadvoice.com id #231 [Feb 22 17:24:43] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:sip.broadvoice.com SIP/2.0 (39) [Feb 22 17:24:43] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK012cd343;rport (64) [Feb 22 17:24:43] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: ;tag=as45a1777c (56) [Feb 22 17:24:43] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (39) [Feb 22 17:24:43] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 42fcb9a2205a226e1657915a20f1c05a@24.123.23.170 (55) [Feb 22 17:24:43] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: CSeq: 108 REGISTER (18) [Feb 22 17:24:43] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Feb 22 17:24:43] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: Max-Forwards: 70 (16) [Feb 22 17:24:43] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Expires: 120 (12) [Feb 22 17:24:43] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Contact: (39) [Feb 22 17:24:43] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Event: registration (19) [Feb 22 17:24:43] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Content-Length: 0 (17) [Feb 22 17:24:43] REGISTER 12 headers, 0 lines [Feb 22 17:24:43] REGISTER attempt 1 to 3173241052@sip.broadvoice.com [Feb 22 17:24:43] 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=z9hG4bK012cd343;rport From: ;tag=as45a1777c To: Call-ID: 42fcb9a2205a226e1657915a20f1c05a@24.123.23.170 CSeq: 108 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: Event: registration Content-Length: 0 --- [Feb 22 17:24:43] DEBUG[14767]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #232 [Feb 22 17:24:43] <--- SIP read from 147.135.12.128:5060 ---> SIP/2.0 200 OK Call-ID: 42fcb9a2205a226e1657915a20f1c05a@24.123.23.170 CSeq: 108 REGISTER From: ;tag=as45a1777c To: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK012cd343 Contact: Expires: 30 Event: registration Content-Length: 0 <-------------> [Feb 22 17:24:43] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 22 17:24:43] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Call-ID: 42fcb9a2205a226e1657915a20f1c05a@24.123.23.170 (55) [Feb 22 17:24:43] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: CSeq: 108 REGISTER (18) [Feb 22 17:24:43] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: From: ;tag=as45a1777c (56) [Feb 22 17:24:43] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: To: (39) [Feb 22 17:24:43] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK012cd343 (58) [Feb 22 17:24:43] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: (39) [Feb 22 17:24:43] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: Expires: 30 (11) [Feb 22 17:24:43] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Event: registration (19) [Feb 22 17:24:43] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (20) [Feb 22 17:24:43] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: (0) [Feb 22 17:24:43] --- (10 headers 0 lines) --- [Feb 22 17:24:43] DEBUG[14767]: chan_sip.c:2061 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #232 [Feb 22 17:24:43] DEBUG[14767]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '42fcb9a2205a226e1657915a20f1c05a@24.123.23.170' of Request 108: Match Not Found [Feb 22 17:24:43] DEBUG[14767]: chan_sip.c:11906 handle_response_register: Registration successful [Feb 22 17:24:43] DEBUG[14767]: chan_sip.c:11909 handle_response_register: Cancelling timeout 231 [Feb 22 17:24:43] Scheduling destruction of SIP dialog '42fcb9a2205a226e1657915a20f1c05a@24.123.23.170' in 32000 ms (Method: REGISTER) [Feb 22 17:24:43] NOTICE[14767]: chan_sip.c:11961 handle_response_register: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 23 s) [Feb 22 17:24:46] <--- SIP read from 192.168.1.86:5060 ---> <-------------> [Feb 22 17:24:46] --- (0 headers 0 lines) Nat keepalive --- [Feb 22 17:24:46] <--- SIP read from 192.168.1.93:5060 ---> BYE sip:526@24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKg14bd4f98f3dd827e49647238af664b04 Call-ID: 524091ef199964d13508e0184586dc1c@24.123.23.170 CSeq: 440470 BYE From: 528 ;tag=81c5c0cdcd26df8f2948fc3e1971f0eb To: ;tag=as449238ce 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: 0 <-------------> [Feb 22 17:24:46] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: BYE sip:526@24.123.23.170 SIP/2.0 (33) [Feb 22 17:24:46] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKg14bd4f98f3dd827e49647238af664b04 (82) [Feb 22 17:24:46] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: Call-ID: 524091ef199964d13508e0184586dc1c@24.123.23.170 (55) [Feb 22 17:24:46] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: CSeq: 440470 BYE (16) [Feb 22 17:24:46] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: 528 ;tag=81c5c0cdcd26df8f2948fc3e1971f0eb (74) [Feb 22 17:24:46] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: To: ;tag=as449238ce (42) [Feb 22 17:24:46] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:24:46] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: Max-Forwards: 70 (16) [Feb 22 17:24:46] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:24:46] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:24:46] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Content-Length: 0 (17) [Feb 22 17:24:46] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: (0) [Feb 22 17:24:46] --- (11 headers 0 lines) --- [Feb 22 17:24:46] <--- Transmitting (no NAT) to 192.168.1.93:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKg14bd4f98f3dd827e49647238af664b04;received=192.168.1.93 From: 528 ;tag=81c5c0cdcd26df8f2948fc3e1971f0eb To: ;tag=as449238ce Call-ID: 524091ef199964d13508e0184586dc1c@24.123.23.170 CSeq: 440470 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> [Feb 22 17:24:46] DEBUG[14767]: chan_sip.c:14631 sipsock_read: Invalid SIP message - rejected , no callid, len 508 quit[Feb 22 17:24:48] <--- SIP read from 68.58.36.157:5060 ---> <-------------> [Feb 22 17:24:48] --- (0 headers 0 lines) Nat keepalive --- [Feb 22 17:24:48] DEBUG[14767]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '000ff78d-ebb20002-237ee55f-6d451a9c@192.168.1.165' [Feb 22 17:24:48] DEBUG[14767]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 000ff78d-ebb20002-237ee55f-6d451a9c@192.168.1.165 [Feb 22 17:24:48] Really destroying SIP dialog '000ff78d-ebb20002-237ee55f-6d451a9c@192.168.1.165' Method: REGISTER exit[Feb 22 17:24:49] DEBUG[14767]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '000ff78d-ebb20004-75d3ddb3-71096c5c@192.168.1.165' [Feb 22 17:24:49] DEBUG[14767]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 000ff78d-ebb20004-75d3ddb3-71096c5c@192.168.1.165 [Feb 22 17:24:49] Really destroying SIP dialog '000ff78d-ebb20004-75d3ddb3-71096c5c@192.168.1.165' Method: REGISTER No such command 'exit' (type 'help' for help) *CLI> [Feb 22 17:24:50] <--- SIP read from 192.168.1.91:5060 ---> REGISTER sip:24.123.23.170:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.91:5060;rport;branch=z9hG4bK1110176608 From: ;tag=4241711383 To: Call-ID: 2731658924@192.168.1.91 CSeq: 3 REGISTER Contact: ;action=proxy max-forwards: 70 Expires: 60 user-agent: UTSTARCOM F3000/Device ID-F3000_TEST Content-Length: 0 <-------------> [Feb 22 17:24:50] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170:5060 SIP/2.0 (39) [Feb 22 17:24:50] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.91:5060;rport;branch=z9hG4bK1110176608 (65) [Feb 22 17:24:50] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: ;tag=4241711383 (49) [Feb 22 17:24:50] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (32) [Feb 22 17:24:50] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 2731658924@192.168.1.91 (32) [Feb 22 17:24:50] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: CSeq: 3 REGISTER (16) [Feb 22 17:24:50] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: ;action=proxy (49) [Feb 22 17:24:50] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: max-forwards: 70 (16) [Feb 22 17:24:50] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Expires: 60 (11) [Feb 22 17:24:50] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: user-agent: UTSTARCOM F3000/Device ID-F3000_TEST (48) [Feb 22 17:24:50] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Content-Length: 0 (17) [Feb 22 17:24:50] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: (0) [Feb 22 17:24:50] --- (11 headers 0 lines) --- [Feb 22 17:24:50] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 2731658924@192.168.1.91 - REGISTER (No RTP) [Feb 22 17:24:50] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:24:50] Using latest REGISTER request as basis request [Feb 22 17:24:50] Sending to 192.168.1.91 : 5060 (NAT) [Feb 22 17:24:50] <--- Transmitting (NAT) to 192.168.1.91:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.91:5060;branch=z9hG4bK1110176608;received=192.168.1.91;rport=5060 From: ;tag=4241711383 To: Call-ID: 2731658924@192.168.1.91 CSeq: 3 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:24:50] <--- Transmitting (NAT) to 192.168.1.91:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.91:5060;branch=z9hG4bK1110176608;received=192.168.1.91;rport=5060 From: ;tag=4241711383 To: ;tag=as190cd07a Call-ID: 2731658924@192.168.1.91 CSeq: 3 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1a61372c" Content-Length: 0 <------------> [Feb 22 17:24:50] Scheduling destruction of SIP dialog '2731658924@192.168.1.91' in 32000 ms (Method: REGISTER) quit No such command 'quit' (type 'help' for help) *CLI> [Feb 22 17:24:51] DEBUG[14767]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '42fcb9a2205a226e1657915a20f1c05a@24.123.23.170' [Feb 22 17:24:51] DEBUG[14767]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 42fcb9a2205a226e1657915a20f1c05a@24.123.23.170 [Feb 22 17:24:51] Really destroying SIP dialog '42fcb9a2205a226e1657915a20f1c05a@24.123.23.170' Method: REGISTER [Feb 22 17:24:52] <--- SIP read from 192.168.1.91:5060 ---> REGISTER sip:24.123.23.170:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.91:5060;rport;branch=z9hG4bK3408698054 From: ;tag=2228096917 To: Call-ID: 2731658924@192.168.1.91 CSeq: 4 REGISTER Contact: ;action=proxy Authorization: Digest username="550", realm="asterisk", nonce="1a61372c", uri="sip:24.123.23.170:5060", response="9c9cb19687ef4fb5d3aae6374d82aef2", algorithm=MD5 max-forwards: 70 Expires: 60 user-agent: UTSTARCOM F3000/Device ID-F3000_TEST Content-Length: 0 <-------------> [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170:5060 SIP/2.0 (39) [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.91:5060;rport;branch=z9hG4bK3408698054 (65) [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: ;tag=2228096917 (49) [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (32) [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 2731658924@192.168.1.91 (32) [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: CSeq: 4 REGISTER (16) [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: ;action=proxy (49) [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: Authorization: Digest username="550", realm="asterisk", nonce="1a61372c", uri="sip:24.123.23.170:5060", response="9c9cb19687ef4fb5d3aae6374d82aef2", algorithm=MD5 (162) [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: max-forwards: 70 (16) [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Expires: 60 (11) [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: user-agent: UTSTARCOM F3000/Device ID-F3000_TEST (48) [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Content-Length: 0 (17) [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: (0) [Feb 22 17:24:52] --- (12 headers 0 lines) --- [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:24:52] Using latest REGISTER request as basis request [Feb 22 17:24:52] Sending to 192.168.1.91 : 5060 (NAT) [Feb 22 17:24:52] <--- Transmitting (NAT) to 192.168.1.91:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.91:5060;branch=z9hG4bK3408698054;received=192.168.1.91;rport=5060 From: ;tag=2228096917 To: Call-ID: 2731658924@192.168.1.91 CSeq: 4 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:24:52] <--- Transmitting (NAT) to 192.168.1.91:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.91:5060;branch=z9hG4bK3408698054;received=192.168.1.91;rport=5060 From: ;tag=2228096917 To: ;tag=as190cd07a Call-ID: 2731658924@192.168.1.91 CSeq: 4 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Thu, 22 Feb 2007 22:24:52 GMT Content-Length: 0 <------------> [Feb 22 17:24:52] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/550 [Feb 22 17:24:52] Scheduling destruction of SIP dialog '2731658924@192.168.1.91' in 32000 ms (Method: REGISTER) [Feb 22 17:24:52] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 550 [Feb 22 17:24:52] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 550 [Feb 22 17:24:52] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/550 - state 1 (Not in use) [Feb 22 17:24:52] DEBUG[14850]: app_queue.c:546 changethread: Device 'SIP/550' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Feb 22 17:24:52] DEBUG[14851]: manager.c:1931 process_message: Manager received command 'Login' [Feb 22 17:24:52] == Parsing '/etc/asterisk/manager.conf': [Feb 22 17:24:52] DEBUG[14851]: config.c:844 config_text_file_load: Parsing /etc/asterisk/manager.conf [Feb 22 17:24:52] Found [Feb 22 17:24:52] DEBUG[14851]: acl.c:199 ast_append_ha: 127.0.0.1/255.255.255.0/255.255.255.0 appended to acl for peer [Feb 22 17:24:52] DEBUG[14851]: acl.c:213 ast_apply_ha: ##### Testing 127.0.0.1 with 127.0.0.0 [Feb 22 17:24:52] == Manager 'MessageNet' logged on from 127.0.0.1 [Feb 22 17:24:52] DEBUG[14851]: manager.c:1931 process_message: Manager received command 'Command' [Feb 22 17:24:52] DEBUG[14851]: manager.c:1931 process_message: Manager received command 'Command' [Feb 22 17:24:52] DEBUG[14851]: manager.c:1931 process_message: Manager received command 'Logoff' [Feb 22 17:24:52] == Manager 'MessageNet' logged off from 127.0.0.1 [Feb 22 17:24:52] <--- 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=z9hG4bKn1500c747b17548bd40ac69c9d6b97f51 Call-ID: 55ae6647666e@24.123.23.170 CSeq: 64476 REGISTER From: ;tag=6ed4804cafe4ed8d2f792fa158f0d2d1 To: Contact: ;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="50a4fb98", algorithm=MD5, uri="sip:24.123.23.170", username="526", response="f47b7922eb51965b0a9127c7f5f7ae7a" <-------------> [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bKn1500c747b17548bd40ac69c9d6b97f51 (82) [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: Call-ID: 55ae6647666e@24.123.23.170 (35) [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: CSeq: 64476 REGISTER (20) [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: ;tag=6ed4804cafe4ed8d2f792fa158f0d2d1 (66) [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: To: (27) [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="50a4fb98", algorithm=MD5, uri="sip:24.123.23.170", username="526", response="f47b7922eb51965b0a9127c7f5f7ae7a" (167) [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 13: (0) [Feb 22 17:24:52] --- (13 headers 0 lines) --- [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 55ae6647666e@24.123.23.170 - REGISTER (No RTP) [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:24:52] Using latest REGISTER request as basis request [Feb 22 17:24:52] Sending to 192.168.1.66 : 5060 (no NAT) [Feb 22 17:24:52] <--- 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=z9hG4bKn1500c747b17548bd40ac69c9d6b97f51;received=192.168.1.66 From: ;tag=6ed4804cafe4ed8d2f792fa158f0d2d1 To: Call-ID: 55ae6647666e@24.123.23.170 CSeq: 64476 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:24:52] <--- 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=z9hG4bKn1500c747b17548bd40ac69c9d6b97f51;received=192.168.1.66 From: ;tag=6ed4804cafe4ed8d2f792fa158f0d2d1 To: ;tag=as034661f7 Call-ID: 55ae6647666e@24.123.23.170 CSeq: 64476 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="037b1649" Content-Length: 0 <------------> [Feb 22 17:24:52] Scheduling destruction of SIP dialog '55ae6647666e@24.123.23.170' in 32000 ms (Method: REGISTER) [Feb 22 17:24:52] <--- 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=z9hG4bKq60bbb162c1b4d31361e558869364316a CSeq: 64477 REGISTER Call-ID: 55ae6647666e@24.123.23.170 From: ;tag=6ed4804cafe4ed8d2f792fa158f0d2d1 To: Contact: ;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="037b1649", algorithm=MD5, uri="sip:24.123.23.170", username="526", response="32880fcf6404aeba3dd92c829196d7c7" <-------------> [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bKq60bbb162c1b4d31361e558869364316a (82) [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: CSeq: 64477 REGISTER (20) [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: Call-ID: 55ae6647666e@24.123.23.170 (35) [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: ;tag=6ed4804cafe4ed8d2f792fa158f0d2d1 (66) [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: To: (27) [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="037b1649", algorithm=MD5, uri="sip:24.123.23.170", username="526", response="32880fcf6404aeba3dd92c829196d7c7" (167) [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 13: (0) [Feb 22 17:24:52] --- (13 headers 0 lines) --- [Feb 22 17:24:52] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:24:52] Using latest REGISTER request as basis request [Feb 22 17:24:52] Sending to 192.168.1.66 : 5060 (no NAT) [Feb 22 17:24:52] <--- 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=z9hG4bKq60bbb162c1b4d31361e558869364316a;received=192.168.1.66 From: ;tag=6ed4804cafe4ed8d2f792fa158f0d2d1 To: Call-ID: 55ae6647666e@24.123.23.170 CSeq: 64477 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:24:52] <--- 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=z9hG4bKq60bbb162c1b4d31361e558869364316a;received=192.168.1.66 From: ;tag=6ed4804cafe4ed8d2f792fa158f0d2d1 To: ;tag=as034661f7 Call-ID: 55ae6647666e@24.123.23.170 CSeq: 64477 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Thu, 22 Feb 2007 22:24:52 GMT Content-Length: 0 <------------> [Feb 22 17:24:52] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/526 [Feb 22 17:24:52] Scheduling destruction of SIP dialog '55ae6647666e@24.123.23.170' in 32000 ms (Method: REGISTER) [Feb 22 17:24:52] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 526 [Feb 22 17:24:52] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 526 [Feb 22 17:24:52] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/526 - state 1 (Not in use) [Feb 22 17:24:52] DEBUG[14852]: app_queue.c:546 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. [Feb 22 17:24:53] DEBUG[14767]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '000ff78d-ebb20006-6007ecd7-0ae287ec@192.168.1.165' [Feb 22 17:24:53] DEBUG[14767]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 000ff78d-ebb20006-6007ecd7-0ae287ec@192.168.1.165 [Feb 22 17:24:53] Really destroying SIP dialog '000ff78d-ebb20006-6007ecd7-0ae287ec@192.168.1.165' Method: REGISTER [Feb 22 17:24:53] DEBUG[14767]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '55ae664767e0@24.123.23.170' [Feb 22 17:24:53] DEBUG[14767]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 55ae664767e0@24.123.23.170 [Feb 22 17:24:53] Really destroying SIP dialog '55ae664767e0@24.123.23.170' Method: REGISTER [Feb 22 17:24:55] <--- 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=z9hG4bKuc2050e408db5fc0f5e223622e6a27d14 Call-ID: 55ae664696c0@24.123.23.170 CSeq: 58873 REGISTER From: ;tag=28de45c4c1e4d574e735fc13751b3ea3 To: Contact: ;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="03148311", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="77c5afe31f1f0426d1404601b669358f" <-------------> [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.98:5060;branch=z9hG4bKuc2050e408db5fc0f5e223622e6a27d14 (82) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: Call-ID: 55ae664696c0@24.123.23.170 (35) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: CSeq: 58873 REGISTER (20) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: ;tag=28de45c4c1e4d574e735fc13751b3ea3 (66) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: To: (27) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="03148311", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="77c5afe31f1f0426d1404601b669358f" (167) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 13: (0) [Feb 22 17:24:55] --- (13 headers 0 lines) --- [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 55ae664696c0@24.123.23.170 - REGISTER (No RTP) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:24:55] Using latest REGISTER request as basis request [Feb 22 17:24:55] Sending to 192.168.1.98 : 5060 (no NAT) [Feb 22 17:24:55] <--- 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=z9hG4bKuc2050e408db5fc0f5e223622e6a27d14;received=192.168.1.98 From: ;tag=28de45c4c1e4d574e735fc13751b3ea3 To: Call-ID: 55ae664696c0@24.123.23.170 CSeq: 58873 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:24:55] <--- 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=z9hG4bKuc2050e408db5fc0f5e223622e6a27d14;received=192.168.1.98 From: ;tag=28de45c4c1e4d574e735fc13751b3ea3 To: ;tag=as501f7590 Call-ID: 55ae664696c0@24.123.23.170 CSeq: 58873 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="74c261e8" Content-Length: 0 <------------> [Feb 22 17:24:55] Scheduling destruction of SIP dialog '55ae664696c0@24.123.23.170' in 32000 ms (Method: REGISTER) [Feb 22 17:24:55] <--- 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=z9hG4bK56e1e6bc From: To: Call-ID: 000ff78d-ebb20007-74946e6d-3266b768@192.168.1.165 Date: Thu, 22 Feb 2007 22:26:28 GMT CSeq: 49728 REGISTER User-Agent: CSCO/7 Contact: Content-Length: 0 Expires: 60 <-------------> [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK56e1e6bc (58) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: (40) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (38) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 000ff78d-ebb20007-74946e6d-3266b768@192.168.1.165 (58) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Date: Thu, 22 Feb 2007 22:26:28 GMT (35) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: CSeq: 49728 REGISTER (20) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: CSCO/7 (18) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Contact: (37) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Expires: 60 (11) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: (0) [Feb 22 17:24:55] --- (11 headers 0 lines) --- [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 000ff78d-ebb20007-74946e6d-3266b768@192.168.1.165 - REGISTER (No RTP) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:24:55] Using latest REGISTER request as basis request [Feb 22 17:24:55] Sending to 192.168.1.165 : 5060 (no NAT) [Feb 22 17:24:55] <--- 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=z9hG4bK56e1e6bc;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20007-74946e6d-3266b768@192.168.1.165 CSeq: 49728 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:24:55] <--- 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=z9hG4bK56e1e6bc;received=192.168.1.165 From: To: ;tag=as6f405d3d Call-ID: 000ff78d-ebb20007-74946e6d-3266b768@192.168.1.165 CSeq: 49728 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="133f99a6" Content-Length: 0 <------------> [Feb 22 17:24:55] Scheduling destruction of SIP dialog '000ff78d-ebb20007-74946e6d-3266b768@192.168.1.165' in 32000 ms (Method: REGISTER) [Feb 22 17:24:55] <--- 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=z9hG4bKw724ef9ee7ca06dbfaf9ac99913555b72 CSeq: 58874 REGISTER Call-ID: 55ae664696c0@24.123.23.170 From: ;tag=28de45c4c1e4d574e735fc13751b3ea3 To: Contact: ;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="74c261e8", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="b9cd144bff8916fd206948a7bcc6befe" <-------------> [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.98:5060;branch=z9hG4bKw724ef9ee7ca06dbfaf9ac99913555b72 (82) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: CSeq: 58874 REGISTER (20) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: Call-ID: 55ae664696c0@24.123.23.170 (35) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: From: ;tag=28de45c4c1e4d574e735fc13751b3ea3 (66) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: To: (27) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="74c261e8", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="b9cd144bff8916fd206948a7bcc6befe" (167) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 13: (0) [Feb 22 17:24:55] --- (13 headers 0 lines) --- [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:24:55] Using latest REGISTER request as basis request [Feb 22 17:24:55] Sending to 192.168.1.98 : 5060 (no NAT) [Feb 22 17:24:55] <--- 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=z9hG4bKw724ef9ee7ca06dbfaf9ac99913555b72;received=192.168.1.98 From: ;tag=28de45c4c1e4d574e735fc13751b3ea3 To: Call-ID: 55ae664696c0@24.123.23.170 CSeq: 58874 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:24:55] <--- 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=z9hG4bKw724ef9ee7ca06dbfaf9ac99913555b72;received=192.168.1.98 From: ;tag=28de45c4c1e4d574e735fc13751b3ea3 To: ;tag=as501f7590 Call-ID: 55ae664696c0@24.123.23.170 CSeq: 58874 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Thu, 22 Feb 2007 22:24:55 GMT Content-Length: 0 <------------> [Feb 22 17:24:55] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/529 [Feb 22 17:24:55] Scheduling destruction of SIP dialog '55ae664696c0@24.123.23.170' in 32000 ms (Method: REGISTER) [Feb 22 17:24:55] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 529 [Feb 22 17:24:55] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 529 [Feb 22 17:24:55] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/529 - state 1 (Not in use) [Feb 22 17:24:55] DEBUG[14853]: app_queue.c:546 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. [Feb 22 17:24:55] <--- 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=z9hG4bK60aaf8ca From: To: Call-ID: 000ff78d-ebb20007-74946e6d-3266b768@192.168.1.165 Date: Thu, 22 Feb 2007 22:26:29 GMT CSeq: 49729 REGISTER User-Agent: CSCO/7 Contact: Authorization: Digest username="515",realm="asterisk",uri="sip:24.123.23.170",response="cf26d7b650d83e6f1e7a120584af772e",nonce="133f99a6",algorithm=MD5 Content-Length: 0 Expires: 60 <-------------> [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK60aaf8ca (58) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: (40) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (38) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 000ff78d-ebb20007-74946e6d-3266b768@192.168.1.165 (58) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Date: Thu, 22 Feb 2007 22:26:29 GMT (35) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: CSeq: 49729 REGISTER (20) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: CSCO/7 (18) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Contact: (37) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Authorization: Digest username="515",realm="asterisk",uri="sip:24.123.23.170",response="cf26d7b650d83e6f1e7a120584af772e",nonce="133f99a6",algorithm=MD5 (152) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Content-Length: 0 (17) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Expires: 60 (11) [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: (0) [Feb 22 17:24:55] --- (12 headers 0 lines) --- [Feb 22 17:24:55] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:24:55] Using latest REGISTER request as basis request [Feb 22 17:24:55] Sending to 192.168.1.165 : 5060 (no NAT) [Feb 22 17:24:55] <--- 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=z9hG4bK60aaf8ca;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20007-74946e6d-3266b768@192.168.1.165 CSeq: 49729 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:24:55] <--- 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=z9hG4bK60aaf8ca;received=192.168.1.165 From: To: ;tag=as6f405d3d Call-ID: 000ff78d-ebb20007-74946e6d-3266b768@192.168.1.165 CSeq: 49729 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Thu, 22 Feb 2007 22:24:55 GMT Content-Length: 0 <------------> [Feb 22 17:24:55] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/515 [Feb 22 17:24:55] Scheduling destruction of SIP dialog '000ff78d-ebb20007-74946e6d-3266b768@192.168.1.165' in 32000 ms (Method: REGISTER) [Feb 22 17:24:55] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 515 [Feb 22 17:24:55] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 515 [Feb 22 17:24:55] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/515 - state 1 (Not in use) [Feb 22 17:24:55] DEBUG[14854]: app_queue.c:546 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. [Feb 22 17:24:56] <--- SIP read from 192.168.1.86:5062 ---> <-------------> [Feb 22 17:24:56] --- (0 headers 0 lines) Nat keepalive --- [Feb 22 17:24:58] DEBUG[14767]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '55ae66469f7f@24.123.23.170' [Feb 22 17:24:58] DEBUG[14767]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 55ae66469f7f@24.123.23.170 [Feb 22 17:24:58] Really destroying SIP dialog '55ae66469f7f@24.123.23.170' Method: REGISTER [Feb 22 17:24:59] <--- 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=z9hG4bK6cd7cdfb From: To: Call-ID: 000ff78d-ebb20003-42dc795d-3e449dd6@192.168.1.165 Date: Thu, 22 Feb 2007 22:26:33 GMT CSeq: 49707 REGISTER User-Agent: CSCO/7 Contact: Content-Length: 0 Expires: 60 <-------------> [Feb 22 17:24:59] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:24:59] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK6cd7cdfb (58) [Feb 22 17:24:59] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: (40) [Feb 22 17:24:59] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (38) [Feb 22 17:24:59] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 000ff78d-ebb20003-42dc795d-3e449dd6@192.168.1.165 (58) [Feb 22 17:24:59] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Date: Thu, 22 Feb 2007 22:26:33 GMT (35) [Feb 22 17:24:59] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: CSeq: 49707 REGISTER (20) [Feb 22 17:24:59] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: CSCO/7 (18) [Feb 22 17:24:59] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Contact: (37) [Feb 22 17:24:59] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Feb 22 17:24:59] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Expires: 60 (11) [Feb 22 17:24:59] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: (0) [Feb 22 17:24:59] --- (11 headers 0 lines) --- [Feb 22 17:24:59] DEBUG[14767]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 000ff78d-ebb20003-42dc795d-3e449dd6@192.168.1.165 - REGISTER (No RTP) [Feb 22 17:24:59] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:24:59] Using latest REGISTER request as basis request [Feb 22 17:24:59] Sending to 192.168.1.165 : 5060 (no NAT) [Feb 22 17:24:59] <--- 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=z9hG4bK6cd7cdfb;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20003-42dc795d-3e449dd6@192.168.1.165 CSeq: 49707 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:24:59] <--- 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=z9hG4bK6cd7cdfb;received=192.168.1.165 From: To: ;tag=as220b40cb Call-ID: 000ff78d-ebb20003-42dc795d-3e449dd6@192.168.1.165 CSeq: 49707 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1efc34e4" Content-Length: 0 <------------> [Feb 22 17:24:59] Scheduling destruction of SIP dialog '000ff78d-ebb20003-42dc795d-3e449dd6@192.168.1.165' in 32000 ms (Method: REGISTER) [Feb 22 17:24:59] <--- 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=z9hG4bK4d70dbcd From: To: Call-ID: 000ff78d-ebb20003-42dc795d-3e449dd6@192.168.1.165 Date: Thu, 22 Feb 2007 22:26:33 GMT CSeq: 49708 REGISTER User-Agent: CSCO/7 Contact: Authorization: Digest username="511",realm="asterisk",uri="sip:24.123.23.170",response="a3aa1a8c6bfcd59ab9fa6c979358caa5",nonce="1efc34e4",algorithm=MD5 Content-Length: 0 Expires: 60 <-------------> [Feb 22 17:24:59] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Feb 22 17:24:59] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK4d70dbcd (58) [Feb 22 17:24:59] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 2: From: (40) [Feb 22 17:24:59] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 3: To: (38) [Feb 22 17:24:59] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 000ff78d-ebb20003-42dc795d-3e449dd6@192.168.1.165 (58) [Feb 22 17:24:59] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 5: Date: Thu, 22 Feb 2007 22:26:33 GMT (35) [Feb 22 17:24:59] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 6: CSeq: 49708 REGISTER (20) [Feb 22 17:24:59] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 7: User-Agent: CSCO/7 (18) [Feb 22 17:24:59] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 8: Contact: (37) [Feb 22 17:24:59] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 9: Authorization: Digest username="511",realm="asterisk",uri="sip:24.123.23.170",response="a3aa1a8c6bfcd59ab9fa6c979358caa5",nonce="1efc34e4",algorithm=MD5 (152) [Feb 22 17:24:59] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 10: Content-Length: 0 (17) [Feb 22 17:24:59] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 11: Expires: 60 (11) [Feb 22 17:24:59] DEBUG[14767]: chan_sip.c:4527 parse_request: Header 12: (0) [Feb 22 17:24:59] --- (12 headers 0 lines) --- [Feb 22 17:24:59] DEBUG[14767]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 22 17:24:59] Using latest REGISTER request as basis request [Feb 22 17:24:59] Sending to 192.168.1.165 : 5060 (no NAT) [Feb 22 17:24:59] <--- 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=z9hG4bK4d70dbcd;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20003-42dc795d-3e449dd6@192.168.1.165 CSeq: 49708 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 22 17:24:59] <--- 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=z9hG4bK4d70dbcd;received=192.168.1.165 From: To: ;tag=as220b40cb Call-ID: 000ff78d-ebb20003-42dc795d-3e449dd6@192.168.1.165 CSeq: 49708 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Thu, 22 Feb 2007 22:24:59 GMT Content-Length: 0 <------------> [Feb 22 17:24:59] DEBUG[14767]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/511 [Feb 22 17:24:59] Scheduling destruction of SIP dialog '000ff78d-ebb20003-42dc795d-3e449dd6@192.168.1.165' in 32000 ms (Method: REGISTER) [Feb 22 17:24:59] DEBUG[14741]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 511 [Feb 22 17:24:59] DEBUG[14741]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 511 [Feb 22 17:24:59] DEBUG[14741]: devicestate.c:287 do_state_change: Changing state for SIP/511 - state 1 (Not in use) [Feb 22 17:24:59] DEBUG[14891]: app_queue.c:546 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. [Feb 22 17:25:00] DEBUG[14767]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '55ae66476793@24.123.23.170' [Feb 22 17:25:00] DEBUG[14767]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 55ae66476793@24.123.23.170 [Feb 22 17:25:00] Really destroying SIP dialog '55ae66476793@24.123.23.170' Method: REGISTER [Feb 22 17:25:01] Beginning asterisk shutdown.... [Feb 22 17:25:01] Executing last minute cleanups [Feb 22 17:25:01] == Destroying musiconhold processes [Feb 22 17:25:01] Asterisk cleanly ending (15). [Feb 22 17:25:01] DEBUG[14736]: asterisk.c:1193 quit_handler: Asterisk ending (15).