[Mar 10 12:06:24] Asterisk 1.4.1, Copyright (C) 1999 - 2006 Digium, Inc. and others. [Mar 10 12:06:24] Created by Mark Spencer [Mar 10 12:06:24] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. [Mar 10 12:06:24] This is free software, with components licensed under the GNU General Public [Mar 10 12:06:24] License version 2 and other licenses; you are welcome to redistribute it under [Mar 10 12:06:24] certain conditions. Type 'core show license' for details. [Mar 10 12:06:24] ========================================================================= [Mar 10 12:06:24] == Parsing '/etc/asterisk/asterisk.conf': Parsing /etc/asterisk/asterisk.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] == Parsing '/etc/asterisk/extconfig.conf': Parsing /etc/asterisk/extconfig.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] == Parsing '/etc/asterisk/logger.conf': Parsing /etc/asterisk/logger.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] Asterisk Event Logger Started /var/log/asterisk/event_log [Mar 10 12:06:24] Asterisk Dynamic Loader Starting: [Mar 10 12:06:24] == Parsing '/etc/asterisk/modules.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/modules.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] == Parsing '/etc/asterisk/dnsmgr.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/dnsmgr.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] == Manager registered action Ping [Mar 10 12:06:24] == Manager registered action Events [Mar 10 12:06:24] == Manager registered action Logoff [Mar 10 12:06:24] == Manager registered action Hangup [Mar 10 12:06:24] == Manager registered action Status [Mar 10 12:06:24] == Manager registered action Setvar [Mar 10 12:06:24] == Manager registered action Getvar [Mar 10 12:06:24] == Manager registered action GetConfig [Mar 10 12:06:24] == Manager registered action UpdateConfig [Mar 10 12:06:24] == Manager registered action Redirect [Mar 10 12:06:24] == Manager registered action Originate [Mar 10 12:06:24] == Manager registered action Command [Mar 10 12:06:24] == Manager registered action ExtensionState [Mar 10 12:06:24] == Manager registered action AbsoluteTimeout [Mar 10 12:06:24] == Manager registered action MailboxStatus [Mar 10 12:06:24] == Manager registered action MailboxCount [Mar 10 12:06:24] == Manager registered action ListCommands [Mar 10 12:06:24] == Manager registered action UserEvent [Mar 10 12:06:24] == Manager registered action WaitEvent [Mar 10 12:06:24] == Parsing '/etc/asterisk/manager.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/manager.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] Asterisk Management interface listening on port 5038 [Mar 10 12:06:24] == Parsing '/etc/asterisk/cdr.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/cdr.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] NOTICE[18522]: cdr.c:1093 do_reload: CDR simple logging enabled. [Mar 10 12:06:24] == Parsing '/etc/asterisk/rtp.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/rtp.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] == RTP Allocating from port range 10000 -> 20000 [Mar 10 12:06:24] == UDPTL allocating from port range 4500 -> 4999 [Mar 10 12:06:24] Asterisk PBX Core Initializing [Mar 10 12:06:24] Registering builtin applications: [Mar 10 12:06:24] [Answer] [Mar 10 12:06:24] == Registered application 'Answer' [Mar 10 12:06:24] [BackGround] [Mar 10 12:06:24] == Registered application 'BackGround' [Mar 10 12:06:24] [Busy] [Mar 10 12:06:24] == Registered application 'Busy' [Mar 10 12:06:24] [Congestion] [Mar 10 12:06:24] == Registered application 'Congestion' [Mar 10 12:06:24] [Goto] [Mar 10 12:06:24] == Registered application 'Goto' [Mar 10 12:06:24] [GotoIf] [Mar 10 12:06:24] == Registered application 'GotoIf' [Mar 10 12:06:24] [GotoIfTime] [Mar 10 12:06:24] == Registered application 'GotoIfTime' [Mar 10 12:06:24] [ExecIfTime] [Mar 10 12:06:24] == Registered application 'ExecIfTime' [Mar 10 12:06:24] [Hangup] [Mar 10 12:06:24] == Registered application 'Hangup' [Mar 10 12:06:24] [NoOp] [Mar 10 12:06:24] == Registered application 'NoOp' [Mar 10 12:06:24] [Progress] [Mar 10 12:06:24] == Registered application 'Progress' [Mar 10 12:06:24] [ResetCDR] [Mar 10 12:06:24] == Registered application 'ResetCDR' [Mar 10 12:06:24] [Ringing] [Mar 10 12:06:24] == Registered application 'Ringing' [Mar 10 12:06:24] [SayNumber] [Mar 10 12:06:24] == Registered application 'SayNumber' [Mar 10 12:06:24] [SayDigits] [Mar 10 12:06:24] == Registered application 'SayDigits' [Mar 10 12:06:24] [SayAlpha] [Mar 10 12:06:24] == Registered application 'SayAlpha' [Mar 10 12:06:24] [SayPhonetic] [Mar 10 12:06:24] == Registered application 'SayPhonetic' [Mar 10 12:06:24] [SetAMAFlags] [Mar 10 12:06:24] == Registered application 'SetAMAFlags' [Mar 10 12:06:24] [SetGlobalVar] [Mar 10 12:06:24] == Registered application 'SetGlobalVar' [Mar 10 12:06:24] [Set] [Mar 10 12:06:24] == Registered application 'Set' [Mar 10 12:06:24] [ImportVar] [Mar 10 12:06:24] == Registered application 'ImportVar' [Mar 10 12:06:24] [Wait] [Mar 10 12:06:24] == Registered application 'Wait' [Mar 10 12:06:24] [WaitExten] [Mar 10 12:06:24] == Registered application 'WaitExten' [Mar 10 12:06:24] == Manager registered action DBGet [Mar 10 12:06:24] == Manager registered action DBPut [Mar 10 12:06:24] == Parsing '/etc/asterisk/enum.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/enum.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] Asterisk Dynamic Loader Starting: [Mar 10 12:06:24] == Parsing '/etc/asterisk/modules.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/modules.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] NOTICE[18522]: loader.c:799 load_modules: 145 modules will be loaded. [Mar 10 12:06:24] == Registered application 'MusicOnHold' [Mar 10 12:06:24] == Registered application 'WaitMusicOnHold' [Mar 10 12:06:24] == Registered application 'SetMusicOnHold' [Mar 10 12:06:24] == Registered application 'StartMusicOnHold' [Mar 10 12:06:24] == Registered application 'StopMusicOnHold' [Mar 10 12:06:24] == Parsing '/etc/asterisk/musiconhold.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/musiconhold.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] WARNING[18522]: res_musiconhold.c:1056 load_moh_classes: The old musiconhold.conf syntax has been deprecated! Please refer to the sample configuration for information on the new syntax. [Mar 10 12:06:24] res_musiconhold.so => (Music On Hold Resource) [Mar 10 12:06:24] == Registered application 'Monitor' [Mar 10 12:06:24] == Registered application 'StopMonitor' [Mar 10 12:06:24] == Registered application 'ChangeMonitor' [Mar 10 12:06:24] == Registered application 'PauseMonitor' [Mar 10 12:06:24] == Registered application 'UnpauseMonitor' [Mar 10 12:06:24] == Manager registered action Monitor [Mar 10 12:06:24] == Manager registered action StopMonitor [Mar 10 12:06:24] == Manager registered action ChangeMonitor [Mar 10 12:06:24] == Manager registered action PauseMonitor [Mar 10 12:06:24] == Manager registered action UnpauseMonitor [Mar 10 12:06:24] res_monitor.so => (Call Monitoring Resource) [Mar 10 12:06:24] == Parsing '/etc/asterisk/indications.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/indications.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] -- Registered indication country 'cl' [Mar 10 12:06:24] -- Registered indication country 'tw' [Mar 10 12:06:24] -- Registered indication country 'us' [Mar 10 12:06:24] -- Registered indication country 'au' [Mar 10 12:06:24] -- Registered indication country 'fr' [Mar 10 12:06:24] -- Registered indication country 'de' [Mar 10 12:06:24] -- Registered indication country 'nl' [Mar 10 12:06:24] -- Registered indication country 'uk' [Mar 10 12:06:24] -- Registered indication country 'fi' [Mar 10 12:06:24] -- Registered indication country 'no' [Mar 10 12:06:24] -- Registered indication country 'br' [Mar 10 12:06:24] -- Registered indication country 'za' [Mar 10 12:06:24] -- Registered indication country 'it' [Mar 10 12:06:24] -- Registered indication country 'us-o' [Mar 10 12:06:24] -- Registered indication country 'gr' [Mar 10 12:06:24] -- Registered indication country 'ru' [Mar 10 12:06:24] -- Registered indication country 'nz' [Mar 10 12:06:24] -- Setting default indication country to 'us' [Mar 10 12:06:24] == Registered application 'PlayTones' [Mar 10 12:06:24] == Registered application 'StopPlayTones' [Mar 10 12:06:24] res_indications.so => (Indications Resource) [Mar 10 12:06:24] == Parsing '/etc/asterisk/features.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/features.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] DEBUG[18522]: pbx.c:3853 __ast_context_create: Registered context 'parkedcalls' [Mar 10 12:06:24] -- Registered extension context 'parkedcalls' [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '700' priority 1 to parkedcalls [Mar 10 12:06:24] -- Added extension '700' priority 1 to parkedcalls [Mar 10 12:06:24] DEBUG[18522]: res_features.c:277 notify_metermaids: Notification of state change to metermaids 700@parkedcalls [Mar 10 12:06:24] DEBUG[18522]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel park:700@parkedcalls [Mar 10 12:06:24] DEBUG[18527]: devicestate.c:157 ast_device_state: Checking if I can find provider for "park" - number: 700@parkedcalls [Mar 10 12:06:24] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for park:700@parkedcalls - state 4 (Invalid) [Mar 10 12:06:24] == Registered application 'ParkedCall' [Mar 10 12:06:24] == Registered application 'Park' [Mar 10 12:06:24] == Manager registered action ParkedCalls [Mar 10 12:06:24] == Manager registered action Park [Mar 10 12:06:24] res_features.so => (Call Features Resource) [Mar 10 12:06:24] -- Loaded PUBLIC key 'freeworlddialup' [Mar 10 12:06:24] DEBUG[18522]: res_crypto.c:259 try_load_key: Key 'freeworlddialup' loaded OK [Mar 10 12:06:24] -- Loaded PUBLIC key 'iaxtel' [Mar 10 12:06:24] DEBUG[18522]: res_crypto.c:259 try_load_key: Key 'iaxtel' loaded OK [Mar 10 12:06:24] res_crypto.so => (Cryptographic Digital Signatures) [Mar 10 12:06:24] == Parsing '/etc/asterisk/adsi.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/adsi.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] res_adsi.so => (ADSI Resource) [Mar 10 12:06:24] res_speech.so => (Generic Speech Recognition API) [Mar 10 12:06:24] NOTICE[18522]: res_smdi.c:539 smdi_load: Unable to load config smdi.conf: SMDI disabled [Mar 10 12:06:24] WARNING[18522]: res_smdi.c:722 load_module: No SMDI interfaces are available to listen on, not starting SDMI listener. [Mar 10 12:06:24] == Registered application 'SetCDRUserField' [Mar 10 12:06:24] == Registered application 'AppendCDRUserField' [Mar 10 12:06:24] == Manager registered action SetCDRUserField [Mar 10 12:06:24] app_setcdruserfield.so => (CDR user field apps) [Mar 10 12:06:24] == Parsing '/etc/asterisk/queues.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/queues.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] == Registered application 'Queue' [Mar 10 12:06:24] == Registered application 'AddQueueMember' [Mar 10 12:06:24] == Registered application 'RemoveQueueMember' [Mar 10 12:06:24] == Registered application 'PauseQueueMember' [Mar 10 12:06:24] == Registered application 'UnpauseQueueMember' [Mar 10 12:06:24] == Registered application 'QueueLog' [Mar 10 12:06:24] == Manager registered action Queues [Mar 10 12:06:24] == Manager registered action QueueStatus [Mar 10 12:06:24] == Manager registered action QueueAdd [Mar 10 12:06:24] == Manager registered action QueueRemove [Mar 10 12:06:24] == Manager registered action QueuePause [Mar 10 12:06:24] == Registered custom function QUEUEAGENTCOUNT [Mar 10 12:06:24] == Registered custom function QUEUE_MEMBER_COUNT [Mar 10 12:06:24] == Registered custom function QUEUE_MEMBER_LIST [Mar 10 12:06:24] == Registered custom function QUEUE_WAITING_COUNT [Mar 10 12:06:24] app_queue.so => (True Call Queueing) [Mar 10 12:06:24] pbx_spool.so => (Outgoing Spool Support) [Mar 10 12:06:24] == Registered application 'Dictate' [Mar 10 12:06:24] app_dictate.so => (Virtual Dictation Machine) [Mar 10 12:06:24] == Registered application 'DISA' [Mar 10 12:06:24] app_disa.so => (DISA (Direct Inward System Access) Application) [Mar 10 12:06:24] == Registered custom function DB [Mar 10 12:06:24] == Registered custom function DB_EXISTS [Mar 10 12:06:24] == Registered custom function DB_DELETE [Mar 10 12:06:24] func_db.so => (Database (astdb) related dialplan functions) [Mar 10 12:06:24] == Registered custom function MUSICCLASS [Mar 10 12:06:24] func_moh.so => (Music-on-hold dialplan function) [Mar 10 12:06:24] == Registered application 'ChanIsAvail' [Mar 10 12:06:24] app_chanisavail.so => (Check channel availability) [Mar 10 12:06:24] == Parsing '/etc/asterisk/cdr_custom.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/cdr_custom.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] cdr_custom.so => (Customizable Comma Separated Values CDR Backend) [Mar 10 12:06:24] == Parsing '/etc/asterisk/codecs.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/codecs.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] -- codec_alaw: using generic PLC [Mar 10 12:06:24] WARNING[18522]: translate.c:675 __ast_register_translator: plc_samples 160 format 6 [Mar 10 12:06:24] == Registered translator 'alawtolin' from format alaw to slin, cost 1 [Mar 10 12:06:24] DEBUG[18522]: translate.c:425 rebuild_matrix: Resetting translation matrix [Mar 10 12:06:24] == Registered translator 'lintoalaw' from format slin to alaw, cost 1 [Mar 10 12:06:24] DEBUG[18522]: translate.c:425 rebuild_matrix: Resetting translation matrix [Mar 10 12:06:24] codec_alaw.so => (A-law Coder/Decoder) [Mar 10 12:06:24] == Registered custom function MD5 [Mar 10 12:06:24] == Registered custom function CHECK_MD5 [Mar 10 12:06:24] func_md5.so => (MD5 digest dialplan functions) [Mar 10 12:06:24] == Registered custom function SHA1 [Mar 10 12:06:24] func_sha1.so => (SHA-1 computation dialplan function) [Mar 10 12:06:24] == Registered custom function GLOBAL [Mar 10 12:06:24] func_global.so => (Global variable dialplan functions) [Mar 10 12:06:24] == Registered file format ogg_vorbis, extension(s) ogg [Mar 10 12:06:24] format_ogg_vorbis.so => (OGG/Vorbis audio) [Mar 10 12:06:24] == Registered application 'ZapRAS' [Mar 10 12:06:24] app_zapras.so => (Zap RAS Application) [Mar 10 12:06:24] == Registered file format pcm, extension(s) pcm|ulaw|ul|mu [Mar 10 12:06:24] == Registered file format alaw, extension(s) alaw|al [Mar 10 12:06:24] == Registered file format au, extension(s) au [Mar 10 12:06:24] == Registered file format g722, extension(s) g722 [Mar 10 12:06:24] format_pcm.so => (Raw/Sun uLaw/ALaw 8KHz Audio support (PCM,PCMA,AU) and G.722 16Khz Audio Support) [Mar 10 12:06:24] == Registered file format g726-40, extension(s) g726-40 [Mar 10 12:06:24] == Registered file format g726-32, extension(s) g726-32 [Mar 10 12:06:24] == Registered file format g726-24, extension(s) g726-24 [Mar 10 12:06:24] == Registered file format g726-16, extension(s) g726-16 [Mar 10 12:06:24] format_g726.so => (Raw G.726 (16/24/32/40kbps) data) [Mar 10 12:06:24] == Registered custom function GROUP_COUNT [Mar 10 12:06:24] == Registered custom function GROUP_MATCH_COUNT [Mar 10 12:06:24] == Registered custom function GROUP_LIST [Mar 10 12:06:24] == Registered custom function GROUP [Mar 10 12:06:24] func_groupcount.so => (Channel group dialplan functions) [Mar 10 12:06:24] == Registered application 'Page' [Mar 10 12:06:24] app_page.so => (Page Multiple Phones) [Mar 10 12:06:24] == Registered application 'Random' [Mar 10 12:06:24] app_random.so => (Random goto) [Mar 10 12:06:24] == Registered file format gsm, extension(s) gsm [Mar 10 12:06:24] format_gsm.so => (Raw GSM data) [Mar 10 12:06:24] == Registered translator 'ilbctolin' from format ilbc to slin, cost 2 [Mar 10 12:06:24] DEBUG[18522]: translate.c:425 rebuild_matrix: Resetting translation matrix [Mar 10 12:06:24] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:24] == Registered translator 'lintoilbc' from format slin to ilbc, cost 14 [Mar 10 12:06:24] DEBUG[18522]: translate.c:425 rebuild_matrix: Resetting translation matrix [Mar 10 12:06:24] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Mar 10 12:06:24] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:24] codec_ilbc.so => (iLBC Coder/Decoder) [Mar 10 12:06:24] NOTICE[18522]: pbx_ael.c:3910 pbx_load_module: Starting AEL load process. [Mar 10 12:06:24] NOTICE[18522]: pbx_ael.c:3917 pbx_load_module: AEL load process: calculated config file name '/etc/asterisk/extensions.ael'. [Mar 10 12:06:24] NOTICE[18522]: pbx_ael.c:3925 pbx_load_module: AEL load process: parsed config file name '/etc/asterisk/extensions.ael'. [Mar 10 12:06:24] NOTICE[18522]: pbx_ael.c:3928 pbx_load_module: AEL load process: checked config file name '/etc/asterisk/extensions.ael'. [Mar 10 12:06:24] DEBUG[18522]: pbx.c:3853 __ast_context_create: Registered context 'macro-std-exten-ael' [Mar 10 12:06:24] -- Registered extension context 'macro-std-exten-ael' [Mar 10 12:06:24] DEBUG[18522]: pbx.c:3853 __ast_context_create: Registered context 'ael-demo' [Mar 10 12:06:24] -- Registered extension context 'ael-demo' [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 1 to macro-std-exten-ael [Mar 10 12:06:24] -- Added extension 's' priority 1 to macro-std-exten-ael [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 2 to macro-std-exten-ael [Mar 10 12:06:24] -- Added extension 's' priority 2 to macro-std-exten-ael [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 3 to macro-std-exten-ael [Mar 10 12:06:24] -- Added extension 's' priority 3 to macro-std-exten-ael [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 4 to macro-std-exten-ael [Mar 10 12:06:24] -- Added extension 's' priority 4 to macro-std-exten-ael [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 5 to macro-std-exten-ael [Mar 10 12:06:24] -- Added extension 's' priority 5 to macro-std-exten-ael [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'a' priority 1 to macro-std-exten-ael [Mar 10 12:06:24] -- Added extension 'a' priority 1 to macro-std-exten-ael [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'a' priority 2 to macro-std-exten-ael [Mar 10 12:06:24] -- Added extension 'a' priority 2 to macro-std-exten-ael [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'a' priority 3 to macro-std-exten-ael [Mar 10 12:06:24] -- Added extension 'a' priority 3 to macro-std-exten-ael [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_sw-1-.' priority 1 to macro-std-exten-ael [Mar 10 12:06:24] -- Added extension '_sw-1-.' priority 1 to macro-std-exten-ael [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_sw-1-.' priority 2 to macro-std-exten-ael [Mar 10 12:06:24] -- Added extension '_sw-1-.' priority 2 to macro-std-exten-ael [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'sw-1-BUSY' priority 1 to macro-std-exten-ael [Mar 10 12:06:24] -- Added extension 'sw-1-BUSY' priority 1 to macro-std-exten-ael [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'sw-1-BUSY' priority 2 to macro-std-exten-ael [Mar 10 12:06:24] -- Added extension 'sw-1-BUSY' priority 2 to macro-std-exten-ael [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 1 to ael-demo [Mar 10 12:06:24] -- Added extension 's' priority 1 to ael-demo [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 2 to ael-demo [Mar 10 12:06:24] -- Added extension 's' priority 2 to ael-demo [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 3 to ael-demo [Mar 10 12:06:24] -- Added extension 's' priority 3 to ael-demo [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 4 to ael-demo [Mar 10 12:06:24] -- Added extension 's' priority 4 to ael-demo [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 5 to ael-demo [Mar 10 12:06:24] -- Added extension 's' priority 5 to ael-demo [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 6 to ael-demo [Mar 10 12:06:24] -- Added extension 's' priority 6 to ael-demo [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 7 to ael-demo [Mar 10 12:06:24] -- Added extension 's' priority 7 to ael-demo [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 8 to ael-demo [Mar 10 12:06:24] -- Added extension 's' priority 8 to ael-demo [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 9 to ael-demo [Mar 10 12:06:24] -- Added extension 's' priority 9 to ael-demo [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 10 to ael-demo [Mar 10 12:06:24] -- Added extension 's' priority 10 to ael-demo [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 11 to ael-demo [Mar 10 12:06:24] -- Added extension 's' priority 11 to ael-demo [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 12 to ael-demo [Mar 10 12:06:24] -- Added extension 's' priority 12 to ael-demo [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '2' priority 1 to ael-demo [Mar 10 12:06:24] -- Added extension '2' priority 1 to ael-demo [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '2' priority 2 to ael-demo [Mar 10 12:06:24] -- Added extension '2' priority 2 to ael-demo [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '3' priority 1 to ael-demo [Mar 10 12:06:24] -- Added extension '3' priority 1 to ael-demo [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '3' priority 2 to ael-demo [Mar 10 12:06:24] -- Added extension '3' priority 2 to ael-demo [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '500' priority 1 to ael-demo [Mar 10 12:06:24] -- Added extension '500' priority 1 to ael-demo [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '500' priority 2 to ael-demo [Mar 10 12:06:24] -- Added extension '500' priority 2 to ael-demo [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '500' priority 3 to ael-demo [Mar 10 12:06:24] -- Added extension '500' priority 3 to ael-demo [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '500' priority 4 to ael-demo [Mar 10 12:06:24] -- Added extension '500' priority 4 to ael-demo [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '600' priority 1 to ael-demo [Mar 10 12:06:24] -- Added extension '600' priority 1 to ael-demo [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '600' priority 2 to ael-demo [Mar 10 12:06:24] -- Added extension '600' priority 2 to ael-demo [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '600' priority 3 to ael-demo [Mar 10 12:06:24] -- Added extension '600' priority 3 to ael-demo [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '600' priority 4 to ael-demo [Mar 10 12:06:24] -- Added extension '600' priority 4 to ael-demo [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_1234' priority 1 to ael-demo [Mar 10 12:06:24] -- Added extension '_1234' priority 1 to ael-demo [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '#' priority 1 to ael-demo [Mar 10 12:06:24] -- Added extension '#' priority 1 to ael-demo [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '#' priority 2 to ael-demo [Mar 10 12:06:24] -- Added extension '#' priority 2 to ael-demo [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 't' priority 1 to ael-demo [Mar 10 12:06:24] -- Added extension 't' priority 1 to ael-demo [Mar 10 12:06:24] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'i' priority 1 to ael-demo [Mar 10 12:06:24] -- Added extension 'i' priority 1 to ael-demo [Mar 10 12:06:24] NOTICE[18522]: pbx_ael.c:3930 pbx_load_module: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'. [Mar 10 12:06:24] DEBUG[18522]: pbx.c:3928 ast_merge_contexts_and_delete: must remove any reg pbx_ael [Mar 10 12:06:24] DEBUG[18522]: pbx.c:5265 __ast_context_destroy: check ctx parkedcalls res_features [Mar 10 12:06:24] NOTICE[18522]: pbx_ael.c:3933 pbx_load_module: AEL load process: merged config file name '/etc/asterisk/extensions.ael'. [Mar 10 12:06:24] NOTICE[18522]: pbx_ael.c:3936 pbx_load_module: AEL load process: verified config file name '/etc/asterisk/extensions.ael'. [Mar 10 12:06:24] pbx_ael.so => (Asterisk Extension Language Compiler) [Mar 10 12:06:24] == Registered file format g723sf, extension(s) g723|g723sf [Mar 10 12:06:24] format_g723.so => (G.723.1 Simple Timestamp File Format) [Mar 10 12:06:24] == Registered file format g729, extension(s) g729 [Mar 10 12:06:24] format_g729.so => (Raw G729 data) [Mar 10 12:06:24] == Parsing '/etc/asterisk/cdr.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/cdr.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] == Registered application 'SoftHangup' [Mar 10 12:06:24] app_softhangup.so => (Hangs up the requested channel) [Mar 10 12:06:24] == Registered application 'ParkAndAnnounce' [Mar 10 12:06:24] app_parkandannounce.so => (Call Parking and Announce Application) [Mar 10 12:06:24] pbx_loopback.so => (Loopback Switch) [Mar 10 12:06:24] == Registered application 'Read' [Mar 10 12:06:24] app_read.so => (Read Variable Application) [Mar 10 12:06:24] == Registered custom function VMCOUNT [Mar 10 12:06:24] == Registered application 'HasVoicemail' [Mar 10 12:06:24] == Registered application 'HasNewVoicemail' [Mar 10 12:06:24] app_hasnewvoicemail.so => (Indicator for whether a voice mailbox has messages in a given folder.) [Mar 10 12:06:24] == Registered application 'PrivacyManager' [Mar 10 12:06:24] app_privacy.so => (Require phone number to be entered, if no CallerID sent) [Mar 10 12:06:24] == Parsing '/etc/asterisk/alarmreceiver.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/alarmreceiver.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] == Registered application 'AlarmReceiver' [Mar 10 12:06:24] app_alarmreceiver.so => (Alarm Receiver for Asterisk) [Mar 10 12:06:24] == Registered file format sln, extension(s) sln|raw [Mar 10 12:06:24] format_sln.so => (Raw Signed Linear Audio support (SLN)) [Mar 10 12:06:24] == Registered custom function CUT [Mar 10 12:06:24] == Registered custom function SORT [Mar 10 12:06:24] func_cut.so => (Cut out information from a string) [Mar 10 12:06:24] == Registered application 'WaitForSilence' [Mar 10 12:06:24] app_waitforsilence.so => (Wait For Silence) [Mar 10 12:06:24] res_convert.so => (File format conversion CLI command) [Mar 10 12:06:24] == Registered custom function MATH [Mar 10 12:06:24] func_math.so => (Mathematical dialplan function) [Mar 10 12:06:24] == Registered application 'SendURL' [Mar 10 12:06:24] app_url.so => (Send URL Applications) [Mar 10 12:06:24] == Parsing '/etc/asterisk/festival.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/festival.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] == Registered application 'Festival' [Mar 10 12:06:24] app_festival.so => (Simple Festival Interface) [Mar 10 12:06:24] == Parsing '/etc/asterisk/meetme.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/meetme.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] == Parsing '/etc/asterisk/express.demonstration.meetme.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/express.demonstration.meetme.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] == Manager registered action MeetmeMute [Mar 10 12:06:24] == Manager registered action MeetmeUnmute [Mar 10 12:06:24] == Registered application 'MeetMeAdmin' [Mar 10 12:06:24] == Registered application 'MeetMeCount' [Mar 10 12:06:24] == Registered application 'MeetMe' [Mar 10 12:06:24] == Registered application 'SLAStation' [Mar 10 12:06:24] == Registered application 'SLATrunk' [Mar 10 12:06:24] app_meetme.so => (MeetMe conference bridge) [Mar 10 12:06:24] == Registered application 'MixMonitor' [Mar 10 12:06:24] == Registered application 'StopMixMonitor' [Mar 10 12:06:24] app_mixmonitor.so => (Mixed Audio Monitoring Application) [Mar 10 12:06:24] == Registered application 'ICES' [Mar 10 12:06:24] app_ices.so => (Encode and Stream via icecast and ices) [Mar 10 12:06:24] == Registered format 'jpg' (JPEG (Joint Picture Experts Group)) [Mar 10 12:06:24] format_jpeg.so => (JPEG (Joint Picture Experts Group) Image Format) [Mar 10 12:06:24] == Registered application 'StackPop' [Mar 10 12:06:24] == Registered application 'Return' [Mar 10 12:06:24] == Registered application 'GosubIf' [Mar 10 12:06:24] == Registered application 'Gosub' [Mar 10 12:06:24] app_stack.so => (Stack Routines) [Mar 10 12:06:24] WARNING[18522]: app_followme.c:297 reload_followme: No follow me config file (followme.conf), so no follow me [Mar 10 12:06:24] == Parsing '/etc/asterisk/dundi.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/dundi.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] DEBUG[18522]: pbx_dundi.c:408 reset_global_eid: Seeding global EID '00:01:03:dc:49:ec' from 'eth0' [Mar 10 12:06:24] == Using TOS bits 0 [Mar 10 12:06:24] == DUNDi Ready and Listening on 0.0.0.0 port 4520 [Mar 10 12:06:24] == Registered custom function DUNDILOOKUP [Mar 10 12:06:24] pbx_dundi.so => (Distributed Universal Number Discovery (DUNDi)) [Mar 10 12:06:24] == Registered application 'MacroExit' [Mar 10 12:06:24] == Registered application 'MacroIf' [Mar 10 12:06:24] == Registered application 'MacroExclusive' [Mar 10 12:06:24] == Registered application 'Macro' [Mar 10 12:06:24] app_macro.so => (Extension Macros) [Mar 10 12:06:24] == Registered application 'RealTimeUpdate' [Mar 10 12:06:24] == Registered application 'RealTime' [Mar 10 12:06:24] app_realtime.so => (Realtime Data Lookup/Rewrite) [Mar 10 12:06:24] == Registered application 'DBdel' [Mar 10 12:06:24] == Registered application 'DBdeltree' [Mar 10 12:06:24] app_db.so => (Database Access Functions) [Mar 10 12:06:24] DEBUG[18522]: channel.c:524 ast_channel_register: Registered handler for 'Feature' (Feature Proxy Channel Driver) [Mar 10 12:06:24] == Registered channel type 'Feature' (Feature Proxy Channel Driver) [Mar 10 12:06:24] chan_features.so => (Feature Proxy Channel) [Mar 10 12:06:24] == Registered application 'Milliwatt' [Mar 10 12:06:24] app_milliwatt.so => (Digital Milliwatt (mu-law) Test Application) [Mar 10 12:06:24] == Parsing '/etc/asterisk/skinny.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/skinny.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] WARNING[18522]: chan_skinny.c:4471 reload_config: Option 'port' at line 5 of skinny.conf has been deprecated. Please use 'bindport' instead. [Mar 10 12:06:24] == Skinny listening on 0.0.0.0:2000 [Mar 10 12:06:24] DEBUG[18522]: channel.c:524 ast_channel_register: Registered handler for 'Skinny' (Skinny Client Control Protocol (Skinny)) [Mar 10 12:06:24] == Registered channel type 'Skinny' (Skinny Client Control Protocol (Skinny)) [Mar 10 12:06:24] == Registered custom function ENUMLOOKUP [Mar 10 12:06:24] == Registered custom function TXTCIDNAME [Mar 10 12:06:24] func_enum.so => (ENUM related dialplan functions) [Mar 10 12:06:24] == Registered application 'DumpChan' [Mar 10 12:06:24] app_dumpchan.so => (Dump Info About The Calling Channel) [Mar 10 12:06:24] == Parsing '/etc/asterisk/phone.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/phone.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] DEBUG[18522]: channel.c:524 ast_channel_register: Registered handler for 'Phone' (Standard Linux Telephony API Driver) [Mar 10 12:06:24] == Registered channel type 'Phone' (Standard Linux Telephony API Driver) [Mar 10 12:06:24] chan_phone.so => (Linux Telephony API Support) [Mar 10 12:06:24] == Registered application 'TestClient' [Mar 10 12:06:24] == Registered application 'TestServer' [Mar 10 12:06:24] app_test.so => (Interface Test Application) [Mar 10 12:06:24] == Registered custom function CURL [Mar 10 12:06:24] func_curl.so => (Load external URL) [Mar 10 12:06:24] == Registered application 'Morsecode' [Mar 10 12:06:24] app_morsecode.so => (Morse code) [Mar 10 12:06:24] == Registered application 'Log' [Mar 10 12:06:24] == Registered application 'Verbose' [Mar 10 12:06:24] app_verbose.so => (Send verbose output) [Mar 10 12:06:24] == Parsing '/etc/asterisk/cdr_manager.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/cdr_manager.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] cdr_manager.so => (Asterisk Manager Interface CDR Backend) [Mar 10 12:06:24] == Registered application 'SendText' [Mar 10 12:06:24] app_sendtext.so => (Send Text Applications) [Mar 10 12:06:24] == Registered custom function ISNULL [Mar 10 12:06:24] == Registered custom function SET [Mar 10 12:06:24] == Registered custom function EXISTS [Mar 10 12:06:24] == Registered custom function IF [Mar 10 12:06:24] == Registered custom function IFTIME [Mar 10 12:06:24] func_logic.so => (Logical dialplan functions) [Mar 10 12:06:24] == Parsing '/etc/asterisk/codecs.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/codecs.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] -- codec_lpc10: using generic PLC [Mar 10 12:06:24] WARNING[18522]: translate.c:675 __ast_register_translator: plc_samples 180 format 6 [Mar 10 12:06:24] == Registered translator 'lpc10tolin' from format lpc10 to slin, cost 2 [Mar 10 12:06:24] DEBUG[18522]: translate.c:425 rebuild_matrix: Resetting translation matrix [Mar 10 12:06:24] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Mar 10 12:06:24] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:24] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 6 [Mar 10 12:06:24] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:24] == Registered translator 'lintolpc10' from format slin to lpc10, cost 3 [Mar 10 12:06:24] DEBUG[18522]: translate.c:425 rebuild_matrix: Resetting translation matrix [Mar 10 12:06:24] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:24] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Mar 10 12:06:24] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:24] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 6 [Mar 10 12:06:24] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:24] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6 [Mar 10 12:06:24] codec_lpc10.so => (LPC10 2.4kbps Coder/Decoder) [Mar 10 12:06:24] == Registered custom function CDR [Mar 10 12:06:24] func_cdr.so => (CDR dialplan function) [Mar 10 12:06:24] == Registered custom function RAND [Mar 10 12:06:24] func_rand.so => (Random number dialplan function) [Mar 10 12:06:24] == Registered application 'Echo' [Mar 10 12:06:24] app_echo.so => (Simple Echo Application) [Mar 10 12:06:24] == Registered application 'DeadAGI' [Mar 10 12:06:24] == Registered application 'EAGI' [Mar 10 12:06:24] == Registered application 'AGI' [Mar 10 12:06:24] res_agi.so => (Asterisk Gateway Interface (AGI)) [Mar 10 12:06:24] == Registered application 'Record' [Mar 10 12:06:24] app_record.so => (Trivial Record Application) [Mar 10 12:06:24] == Registered application 'ZapBarge' [Mar 10 12:06:24] app_zapbarge.so => (Barge in on Zap channel application) [Mar 10 12:06:24] == Parsing '/etc/asterisk/mgcp.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/mgcp.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] == MGCP Listening on 0.0.0.0:2727 [Mar 10 12:06:24] == Using TOS bits 0 [Mar 10 12:06:24] DEBUG[18522]: channel.c:524 ast_channel_register: Registered handler for 'MGCP' (Media Gateway Control Protocol (MGCP)) [Mar 10 12:06:24] == Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP)) [Mar 10 12:06:24] chan_mgcp.so => (Media Gateway Control Protocol (MGCP)) [Mar 10 12:06:24] == Parsing '/etc/asterisk/codecs.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/codecs.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] -- codec_gsm: using generic PLC [Mar 10 12:06:24] WARNING[18522]: translate.c:675 __ast_register_translator: plc_samples 160 format 6 [Mar 10 12:06:24] == Registered translator 'gsmtolin' from format gsm to slin, cost 1 [Mar 10 12:06:24] DEBUG[18522]: translate.c:425 rebuild_matrix: Resetting translation matrix [Mar 10 12:06:24] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Mar 10 12:06:24] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6 [Mar 10 12:06:24] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from g723 to unknown, via 6 [Mar 10 12:06:24] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:24] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Mar 10 12:06:24] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:24] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 6 [Mar 10 12:06:24] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:24] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6 [Mar 10 12:06:24] == Registered translator 'lintogsm' from format slin to gsm, cost 2 [Mar 10 12:06:24] DEBUG[18522]: translate.c:425 rebuild_matrix: Resetting translation matrix [Mar 10 12:06:24] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Mar 10 12:06:24] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6 [Mar 10 12:06:24] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from g723 to unknown, via 6 [Mar 10 12:06:24] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6 [Mar 10 12:06:24] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:24] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Mar 10 12:06:24] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Mar 10 12:06:24] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:24] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 6 [Mar 10 12:06:24] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Mar 10 12:06:24] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:24] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6 [Mar 10 12:06:24] codec_gsm.so => (GSM Coder/Decoder) [Mar 10 12:06:24] == Registered custom function IAXPEER [Mar 10 12:06:24] == Registered application 'IAX2Provision' [Mar 10 12:06:24] == Manager registered action IAXpeers [Mar 10 12:06:24] == Manager registered action IAXnetstats [Mar 10 12:06:24] == Parsing '/etc/asterisk/iax.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/iax.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] == Parsing '/etc/asterisk/express.demonstration.extensions.iax.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.iax.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] == Parsing '/etc/asterisk/express.demonstration.extensions.iax.605.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.iax.605.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] == Parsing '/etc/asterisk/express.demonstration.extensions.iax.606.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.iax.606.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] == Parsing '/etc/asterisk/express.demonstration.extensions.iax.614.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.iax.614.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] == Parsing '/etc/asterisk/express.demonstration.extensions.iax.616.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.iax.616.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] == Parsing '/etc/asterisk/express.demonstration.extensions.iax.651.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.iax.651.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] == Parsing '/etc/asterisk/express.demonstration.extensions.iax.698.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.iax.698.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] WARNING[18522]: acl.c:312 ast_str2tos: TOS value lowdelay is deprecated. Please see doc/ip-tos.txt for more information. [Mar 10 12:06:24] == Using TOS bits 16 [Mar 10 12:06:24] == Binding IAX2 to '24.123.23.170:4569' [Mar 10 12:06:24] WARNING[18522]: acl.c:312 ast_str2tos: TOS value lowdelay is deprecated. Please see doc/ip-tos.txt for more information. [Mar 10 12:06:24] > doing dnsmgr_lookup for '192.168.1.10' [Mar 10 12:06:24] > doing dnsmgr_lookup for '192.168.1.10' [Mar 10 12:06:24] DEBUG[18522]: db.c:197 ast_db_get: Unable to find key 'dell_8200_to_unifiedpaging' in family 'IAX/Registry' [Mar 10 12:06:24] > doing dnsmgr_lookup for 'switch-1.nufone.net' [Mar 10 12:06:24] > doing dnsmgr_lookup for '192.168.1.170' [Mar 10 12:06:24] > doing dnsmgr_lookup for '192.168.1.159' [Mar 10 12:06:24] DEBUG[18522]: db.c:197 ast_db_get: Unable to find key '605' in family 'IAX/Registry' [Mar 10 12:06:24] -- Seeding '606' at 192.168.1.166:4569 for 60 [Mar 10 12:06:24] DEBUG[18522]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel IAX2/606 [Mar 10 12:06:24] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for IAX2 - 606 [Mar 10 12:06:24] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for IAX2/606 - state 4 (Invalid) [Mar 10 12:06:24] DEBUG[18539]: 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. [Mar 10 12:06:24] -- Seeding '614' at 192.168.1.167:4569 for 60 [Mar 10 12:06:24] DEBUG[18522]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel IAX2/614 [Mar 10 12:06:24] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for IAX2 - 614 [Mar 10 12:06:24] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for IAX2/614 - state 4 (Invalid) [Mar 10 12:06:24] DEBUG[18540]: 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. [Mar 10 12:06:24] -- Seeding '616' at 74.133.32.69:4569 for 60 [Mar 10 12:06:24] DEBUG[18522]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel IAX2/616 [Mar 10 12:06:24] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for IAX2 - 616 [Mar 10 12:06:24] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for IAX2/616 - state 4 (Invalid) [Mar 10 12:06:24] DEBUG[18541]: app_queue.c:546 changethread: Device 'IAX2/616' changed to state '4' (Invalid) but we don't care because they're not a member of any queue. [Mar 10 12:06:24] -- Seeding '651' at 192.168.1.169:4569 for 60 [Mar 10 12:06:24] DEBUG[18522]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel IAX2/651 [Mar 10 12:06:24] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for IAX2 - 651 [Mar 10 12:06:24] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for IAX2/651 - state 4 (Invalid) [Mar 10 12:06:24] DEBUG[18542]: app_queue.c:546 changethread: Device 'IAX2/651' changed to state '4' (Invalid) but we don't care because they're not a member of any queue. [Mar 10 12:06:24] DEBUG[18522]: db.c:197 ast_db_get: Unable to find key '698' in family 'IAX/Registry' [Mar 10 12:06:24] DEBUG[18522]: channel.c:524 ast_channel_register: Registered handler for 'IAX2' (Inter Asterisk eXchange Driver (Ver 2)) [Mar 10 12:06:24] == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2)) [Mar 10 12:06:24] == 10 helper threaads started [Mar 10 12:06:24] == IAX Ready and Listening [Mar 10 12:06:24] == Loaded firmware 'iaxy.bin' [Mar 10 12:06:24] == Parsing '/etc/asterisk/iaxprov.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/iaxprov.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] WARNING[18522]: acl.c:312 ast_str2tos: TOS value lowdelay is deprecated. Please see doc/ip-tos.txt for more information. [Mar 10 12:06:24] -- Loaded provisioning template 'default' [Mar 10 12:06:24] chan_iax2.so => (Inter Asterisk eXchange (Ver 2)) [Mar 10 12:06:24] == Registered application 'Dial' [Mar 10 12:06:24] == Registered application 'RetryDial' [Mar 10 12:06:24] app_dial.so => (Dialing Application) [Mar 10 12:06:24] == Registered application 'ADSIProg' [Mar 10 12:06:24] app_adsiprog.so => (Asterisk ADSI Programming Application) [Mar 10 12:06:24] == Registered custom function REALTIME [Mar 10 12:06:24] func_realtime.so => (Read/Write values from a RealTime repository) [Mar 10 12:06:24] == Registered application 'Exec' [Mar 10 12:06:24] == Registered application 'TryExec' [Mar 10 12:06:24] == Registered application 'ExecIf' [Mar 10 12:06:24] app_exec.so => (Executes dialplan applications) [Mar 10 12:06:24] == Registered file format h264, extension(s) h264 [Mar 10 12:06:24] format_h264.so => (Raw H.264 data) [Mar 10 12:06:24] == Parsing '/etc/asterisk/sip.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/sip.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] == Parsing '/etc/asterisk/express.demonstration.extensions.sip.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] == Parsing '/etc/asterisk/express.demonstration.extensions.sip.520.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.520.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] == Parsing '/etc/asterisk/express.demonstration.extensions.sip.521.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.521.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] == Parsing '/etc/asterisk/express.demonstration.extensions.sip.522.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.522.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] == Parsing '/etc/asterisk/express.demonstration.extensions.sip.523.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.523.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] == Parsing '/etc/asterisk/express.demonstration.extensions.sip.524.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.524.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] == Parsing '/etc/asterisk/express.demonstration.extensions.sip.525.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.525.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] == Parsing '/etc/asterisk/express.demonstration.extensions.sip.526.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.526.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] == Parsing '/etc/asterisk/express.demonstration.extensions.sip.528.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.528.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] == Parsing '/etc/asterisk/express.demonstration.extensions.sip.529.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.529.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] == Parsing '/etc/asterisk/express.demonstration.extensions.sip.540.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.540.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] == Parsing '/etc/asterisk/express.demonstration.extensions.sip.541.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.541.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] == Parsing '/etc/asterisk/express.demonstration.extensions.sip.550.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.550.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] == Parsing '/etc/asterisk/express.demonstration.extensions.sip.551.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.551.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] == Parsing '/etc/asterisk/express.demonstration.extensions.sip.592.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.592.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] == Parsing '/etc/asterisk/express.demonstration.extensions.sip.593.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.593.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] == Parsing '/etc/asterisk/express.demonstration.extensions.sip.594.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.594.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] == Parsing '/etc/asterisk/express.demonstration.extensions.sip.595.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.595.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] == Parsing '/etc/asterisk/express.demonstration.extensions.sip.596.conf': [Mar 10 12:06:24] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.sip.596.conf [Mar 10 12:06:24] Found [Mar 10 12:06:24] WARNING[18522]: chan_sip.c:15418 handle_common_options: insecure=very at line 315 is deprecated; use insecure=port,invite instead [Mar 10 12:06:24] WARNING[18522]: acl.c:182 ast_append_ha: sip.broadvoice.com is not a valid IP [Mar 10 12:06:25] WARNING[18522]: acl.c:182 ast_append_ha: sip.broadvoice.com is not a valid IP [Mar 10 12:06:25] DEBUG[18522]: chan_sip.c:7616 reg_source_db: SIP Seeding peer from astdb: '510' at 510@192.168.1.165:5060 for 60 [Mar 10 12:06:25] DEBUG[18522]: chan_sip.c:7616 reg_source_db: SIP Seeding peer from astdb: '511' at 511@192.168.1.165:5060 for 60 [Mar 10 12:06:25] DEBUG[18522]: chan_sip.c:7616 reg_source_db: SIP Seeding peer from astdb: '512' at 512@192.168.1.165:5060 for 60 [Mar 10 12:06:25] DEBUG[18522]: chan_sip.c:7616 reg_source_db: SIP Seeding peer from astdb: '513' at 513@192.168.1.165:5060 for 60 [Mar 10 12:06:25] DEBUG[18522]: chan_sip.c:7616 reg_source_db: SIP Seeding peer from astdb: '514' at 514@192.168.1.165:5060 for 60 [Mar 10 12:06:25] DEBUG[18522]: chan_sip.c:7616 reg_source_db: SIP Seeding peer from astdb: '515' at 515@192.168.1.165:5060 for 60 [Mar 10 12:06:25] DEBUG[18522]: db.c:197 ast_db_get: Unable to find key '520' in family 'SIP/Registry' [Mar 10 12:06:25] DEBUG[18522]: db.c:197 ast_db_get: Unable to find key '530' in family 'SIP/Registry' [Mar 10 12:06:25] DEBUG[18522]: db.c:197 ast_db_get: Unable to find key '531' in family 'SIP/Registry' [Mar 10 12:06:25] DEBUG[18522]: db.c:197 ast_db_get: Unable to find key '532' in family 'SIP/Registry' [Mar 10 12:06:25] DEBUG[18522]: db.c:197 ast_db_get: Unable to find key '534' in family 'SIP/Registry' [Mar 10 12:06:25] DEBUG[18522]: db.c:197 ast_db_get: Unable to find key '535' in family 'SIP/Registry' [Mar 10 12:06:25] DEBUG[18522]: db.c:197 ast_db_get: Unable to find key '536' in family 'SIP/Registry' [Mar 10 12:06:25] DEBUG[18522]: db.c:197 ast_db_get: Unable to find key '537' in family 'SIP/Registry' [Mar 10 12:06:25] DEBUG[18522]: chan_sip.c:7616 reg_source_db: SIP Seeding peer from astdb: '540' at 540@192.168.1.62:5060 for 3600 [Mar 10 12:06:25] DEBUG[18522]: db.c:197 ast_db_get: Unable to find key '597' in family 'SIP/Registry' [Mar 10 12:06:25] DEBUG[18522]: db.c:197 ast_db_get: Unable to find key '598' in family 'SIP/Registry' [Mar 10 12:06:25] DEBUG[18522]: db.c:197 ast_db_get: Unable to find key '599' in family 'SIP/Registry' [Mar 10 12:06:25] DEBUG[18522]: db.c:197 ast_db_get: Unable to find key '1001' in family 'SIP/Registry' [Mar 10 12:06:25] DEBUG[18522]: db.c:197 ast_db_get: Unable to find key '1002' in family 'SIP/Registry' [Mar 10 12:06:25] DEBUG[18522]: db.c:197 ast_db_get: Unable to find key '1003' in family 'SIP/Registry' [Mar 10 12:06:25] DEBUG[18522]: db.c:197 ast_db_get: Unable to find key '1004' in family 'SIP/Registry' [Mar 10 12:06:25] DEBUG[18522]: db.c:197 ast_db_get: Unable to find key '521' in family 'SIP/Registry' [Mar 10 12:06:25] DEBUG[18522]: chan_sip.c:7616 reg_source_db: SIP Seeding peer from astdb: '522' at 522@192.168.1.95:5060 for 60 [Mar 10 12:06:25] DEBUG[18522]: chan_sip.c:7616 reg_source_db: SIP Seeding peer from astdb: '523' at 523@192.168.1.85:5060 for 60 [Mar 10 12:06:25] DEBUG[18522]: chan_sip.c:7616 reg_source_db: SIP Seeding peer from astdb: '524' at 524@192.168.1.61:5060 for 60 [Mar 10 12:06:25] DEBUG[18522]: chan_sip.c:7616 reg_source_db: SIP Seeding peer from astdb: '525' at 525@192.168.1.76:5060 for 60 [Mar 10 12:06:25] DEBUG[18522]: chan_sip.c:7616 reg_source_db: SIP Seeding peer from astdb: '526' at 526@192.168.1.66:5060 for 60 [Mar 10 12:06:25] DEBUG[18522]: chan_sip.c:7616 reg_source_db: SIP Seeding peer from astdb: '528' at 528@192.168.1.93:5060 for 60 [Mar 10 12:06:25] DEBUG[18522]: chan_sip.c:7616 reg_source_db: SIP Seeding peer from astdb: '529' at 529@192.168.1.98:5060 for 60 [Mar 10 12:06:25] DEBUG[18522]: chan_sip.c:7616 reg_source_db: SIP Seeding peer from astdb: '541' at 541@192.168.1.90:5060 for 3600 [Mar 10 12:06:25] DEBUG[18522]: db.c:197 ast_db_get: Unable to find key '550' in family 'SIP/Registry' [Mar 10 12:06:25] DEBUG[18522]: chan_sip.c:7616 reg_source_db: SIP Seeding peer from astdb: '551' at 551@192.168.1.97:5060 for 60 [Mar 10 12:06:25] DEBUG[18522]: chan_sip.c:7616 reg_source_db: SIP Seeding peer from astdb: '592' at 592@68.58.36.157:5060 for 3600 [Mar 10 12:06:25] DEBUG[18522]: chan_sip.c:7616 reg_source_db: SIP Seeding peer from astdb: '593' at 593@192.168.1.83:5060 for 3600 [Mar 10 12:06:25] DEBUG[18522]: db.c:197 ast_db_get: Unable to find key '594' in family 'SIP/Registry' [Mar 10 12:06:25] DEBUG[18522]: chan_sip.c:7616 reg_source_db: SIP Seeding peer from astdb: '595' at 595@192.168.1.86:5060 for 3600 [Mar 10 12:06:25] DEBUG[18522]: chan_sip.c:7616 reg_source_db: SIP Seeding peer from astdb: '596' at 596@192.168.1.86:5062 for 3600 [Mar 10 12:06:25] == SIP Listening on 24.123.23.170:5060 [Mar 10 12:06:25] == Using SIP TOS: none [Mar 10 12:06:25] == Parsing '/etc/asterisk/sip_notify.conf': [Mar 10 12:06:25] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/sip_notify.conf [Mar 10 12:06:25] Found [Mar 10 12:06:25] DEBUG[18522]: channel.c:524 ast_channel_register: Registered handler for 'SIP' (Session Initiation Protocol (SIP)) [Mar 10 12:06:25] == Registered channel type 'SIP' (Session Initiation Protocol (SIP)) [Mar 10 12:06:25] == Registered application 'SIPDtmfMode' [Mar 10 12:06:25] == Registered application 'SIPAddHeader' [Mar 10 12:06:25] == Registered custom function SIP_HEADER [Mar 10 12:06:25] == Registered custom function SIPPEER [Mar 10 12:06:25] == Registered custom function SIPCHANINFO [Mar 10 12:06:25] == Registered custom function CHECKSIPDOMAIN [Mar 10 12:06:25] == Manager registered action SIPpeers [Mar 10 12:06:25] == Manager registered action SIPshowpeer [Mar 10 12:06:25] chan_sip.so => (Session Initiation Protocol (SIP)) [Mar 10 12:06:25] == Parsing '/etc/asterisk/extensions.conf': [Mar 10 12:06:25] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/extensions.conf [Mar 10 12:06:25] Found [Mar 10 12:06:25] == Parsing '/etc/asterisk/express.demonstration.extensions.meetme.conf': [Mar 10 12:06:25] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.meetme.conf [Mar 10 12:06:25] Found [Mar 10 12:06:25] == Parsing '/etc/asterisk/express.demonstration.dnis.conf': [Mar 10 12:06:25] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/express.demonstration.dnis.conf [Mar 10 12:06:25] Found [Mar 10 12:06:25] == Parsing '/etc/asterisk/express.demonstration.extensions.meetme.conf': [Mar 10 12:06:25] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/express.demonstration.extensions.meetme.conf [Mar 10 12:06:25] Found [Mar 10 12:06:25] == Setting global variable 'CONSOLE' to 'Console/dsp' [Mar 10 12:06:25] == Setting global variable 'IAXINFO' to 'guest' [Mar 10 12:06:25] == Setting global variable 'TRUNK' to 'Zap/g2' [Mar 10 12:06:25] == Setting global variable 'TRUNKMSD' to '1' [Mar 10 12:06:25] == Setting global variable 'DIAL_TIMEOUT' to '20' [Mar 10 12:06:25] == Setting global variable 'SMVOICE_DIAL_TIMEOUT' to '60' [Mar 10 12:06:25] == Setting global variable 'SMVOICE_ANNOUNCE_CALLER' to '1' [Mar 10 12:06:25] == Setting global variable 'SMVOICE_DIAL_LONG_TIMEOUT' to '120' [Mar 10 12:06:25] == Setting global variable 'OPERATOR' to '510' [Mar 10 12:06:25] == Setting global variable 'OPERATOR_TECHNOLOGY' to 'SIP' [Mar 10 12:06:25] == Setting global variable 'SUPPORT' to '216' [Mar 10 12:06:25] == Setting global variable 'SALES' to '217' [Mar 10 12:06:25] == Setting global variable 'INTERCOM' to 'Zap/8' [Mar 10 12:06:25] == Setting global variable 'SMVOICE_ONHOLD' to '' [Mar 10 12:06:25] DEBUG[18522]: pbx.c:3853 __ast_context_create: Registered context 'iaxtel700' [Mar 10 12:06:25] -- Registered extension context 'iaxtel700' [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_91700XXXXXXX' priority 1 to iaxtel700 [Mar 10 12:06:25] -- Added extension '_91700XXXXXXX' priority 1 to iaxtel700 [Mar 10 12:06:25] DEBUG[18522]: pbx.c:3853 __ast_context_create: Registered context 'iaxprovider' [Mar 10 12:06:25] -- Registered extension context 'iaxprovider' [Mar 10 12:06:25] DEBUG[18522]: pbx.c:3853 __ast_context_create: Registered context 'trunkint' [Mar 10 12:06:25] -- Registered extension context 'trunkint' [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_9011.' priority 1 to trunkint [Mar 10 12:06:25] -- Added extension '_9011.' priority 1 to trunkint [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_9011.' priority 2 to trunkint [Mar 10 12:06:25] -- Added extension '_9011.' priority 2 to trunkint [Mar 10 12:06:25] DEBUG[18522]: pbx.c:3853 __ast_context_create: Registered context 'trunkld' [Mar 10 12:06:25] -- Registered extension context 'trunkld' [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_91NXXNXXXXXX' priority 1 to trunkld [Mar 10 12:06:25] -- Added extension '_91NXXNXXXXXX' priority 1 to trunkld [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_91NXXNXXXXXX' priority 2 to trunkld [Mar 10 12:06:25] -- Added extension '_91NXXNXXXXXX' priority 2 to trunkld [Mar 10 12:06:25] DEBUG[18522]: pbx.c:3853 __ast_context_create: Registered context 'trunklocal' [Mar 10 12:06:25] -- Registered extension context 'trunklocal' [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_9NXXXXXX' priority 1 to trunklocal [Mar 10 12:06:25] -- Added extension '_9NXXXXXX' priority 1 to trunklocal [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_9NXXXXXX' priority 2 to trunklocal [Mar 10 12:06:25] -- Added extension '_9NXXXXXX' priority 2 to trunklocal [Mar 10 12:06:25] DEBUG[18522]: pbx.c:3853 __ast_context_create: Registered context 'trunktollfree' [Mar 10 12:06:25] -- Registered extension context 'trunktollfree' [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_91800NXXXXXX' priority 1 to trunktollfree [Mar 10 12:06:25] -- Added extension '_91800NXXXXXX' priority 1 to trunktollfree [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_91800NXXXXXX' priority 2 to trunktollfree [Mar 10 12:06:25] -- Added extension '_91800NXXXXXX' priority 2 to trunktollfree [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_91888NXXXXXX' priority 1 to trunktollfree [Mar 10 12:06:25] -- Added extension '_91888NXXXXXX' priority 1 to trunktollfree [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_91888NXXXXXX' priority 2 to trunktollfree [Mar 10 12:06:25] -- Added extension '_91888NXXXXXX' priority 2 to trunktollfree [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_91877NXXXXXX' priority 1 to trunktollfree [Mar 10 12:06:25] -- Added extension '_91877NXXXXXX' priority 1 to trunktollfree [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_91877NXXXXXX' priority 2 to trunktollfree [Mar 10 12:06:25] -- Added extension '_91877NXXXXXX' priority 2 to trunktollfree [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_91866NXXXXXX' priority 1 to trunktollfree [Mar 10 12:06:25] -- Added extension '_91866NXXXXXX' priority 1 to trunktollfree [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_91866NXXXXXX' priority 2 to trunktollfree [Mar 10 12:06:25] -- Added extension '_91866NXXXXXX' priority 2 to trunktollfree [Mar 10 12:06:25] DEBUG[18522]: pbx.c:3853 __ast_context_create: Registered context 'international' [Mar 10 12:06:25] -- Registered extension context 'international' [Mar 10 12:06:25] -- Including context 'longdistance' in context 'international' [Mar 10 12:06:25] -- Including context 'trunkint' in context 'international' [Mar 10 12:06:25] DEBUG[18522]: pbx.c:3853 __ast_context_create: Registered context 'longdistance' [Mar 10 12:06:25] -- Registered extension context 'longdistance' [Mar 10 12:06:25] -- Including context 'local' in context 'longdistance' [Mar 10 12:06:25] -- Including context 'trunkld' in context 'longdistance' [Mar 10 12:06:25] DEBUG[18522]: pbx.c:3853 __ast_context_create: Registered context 'local' [Mar 10 12:06:25] -- Registered extension context 'local' [Mar 10 12:06:25] -- Including context 'default' in context 'local' [Mar 10 12:06:25] -- Including context 'parkedcalls' in context 'local' [Mar 10 12:06:25] -- Including context 'trunklocal' in context 'local' [Mar 10 12:06:25] -- Including context 'iaxtel700' in context 'local' [Mar 10 12:06:25] -- Including context 'trunktollfree' in context 'local' [Mar 10 12:06:25] -- Including context 'iaxprovider' in context 'local' [Mar 10 12:06:25] DEBUG[18522]: pbx.c:3853 __ast_context_create: Registered context 'macro-stdexten' [Mar 10 12:06:25] -- Registered extension context 'macro-stdexten' [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 1 to macro-stdexten [Mar 10 12:06:25] -- Added extension 's' priority 1 to macro-stdexten [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 2 to macro-stdexten [Mar 10 12:06:25] -- Added extension 's' priority 2 to macro-stdexten [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's-NOANSWER' priority 1 to macro-stdexten [Mar 10 12:06:25] -- Added extension 's-NOANSWER' priority 1 to macro-stdexten [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's-NOANSWER' priority 2 to macro-stdexten [Mar 10 12:06:25] -- Added extension 's-NOANSWER' priority 2 to macro-stdexten [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's-BUSY' priority 1 to macro-stdexten [Mar 10 12:06:25] -- Added extension 's-BUSY' priority 1 to macro-stdexten [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's-BUSY' priority 2 to macro-stdexten [Mar 10 12:06:25] -- Added extension 's-BUSY' priority 2 to macro-stdexten [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_s-.' priority 1 to macro-stdexten [Mar 10 12:06:25] -- Added extension '_s-.' priority 1 to macro-stdexten [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'a' priority 1 to macro-stdexten [Mar 10 12:06:25] -- Added extension 'a' priority 1 to macro-stdexten [Mar 10 12:06:25] DEBUG[18522]: pbx.c:3853 __ast_context_create: Registered context 'demo' [Mar 10 12:06:25] -- Registered extension context 'demo' [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 1 to demo [Mar 10 12:06:25] -- Added extension 's' priority 1 to demo [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 2 to demo [Mar 10 12:06:25] -- Added extension 's' priority 2 to demo [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 3 to demo [Mar 10 12:06:25] -- Added extension 's' priority 3 to demo [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 4 to demo [Mar 10 12:06:25] -- Added extension 's' priority 4 to demo [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 5 to demo [Mar 10 12:06:25] -- Added extension 's' priority 5 to demo [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 6 to demo [Mar 10 12:06:25] -- Added extension 's' priority 6 to demo [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '2' priority 1 to demo [Mar 10 12:06:25] -- Added extension '2' priority 1 to demo [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '2' priority 2 to demo [Mar 10 12:06:25] -- Added extension '2' priority 2 to demo [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '3' priority 1 to demo [Mar 10 12:06:25] -- Added extension '3' priority 1 to demo [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '3' priority 2 to demo [Mar 10 12:06:25] -- Added extension '3' priority 2 to demo [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '1000' priority 1 to demo [Mar 10 12:06:25] -- Added extension '1000' priority 1 to demo [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '1234' priority 1 to demo [Mar 10 12:06:25] -- Added extension '1234' priority 1 to demo [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '1234' priority 2 to demo [Mar 10 12:06:25] -- Added extension '1234' priority 2 to demo [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '1235' priority 1 to demo [Mar 10 12:06:25] -- Added extension '1235' priority 1 to demo [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '1236' priority 1 to demo [Mar 10 12:06:25] -- Added extension '1236' priority 1 to demo [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '1236' priority 2 to demo [Mar 10 12:06:25] -- Added extension '1236' priority 2 to demo [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '#' priority 1 to demo [Mar 10 12:06:25] -- Added extension '#' priority 1 to demo [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '#' priority 2 to demo [Mar 10 12:06:25] -- Added extension '#' priority 2 to demo [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 't' priority 1 to demo [Mar 10 12:06:25] -- Added extension 't' priority 1 to demo [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'i' priority 1 to demo [Mar 10 12:06:25] -- Added extension 'i' priority 1 to demo [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '500' priority 1 to demo [Mar 10 12:06:25] -- Added extension '500' priority 1 to demo [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '500' priority 2 to demo [Mar 10 12:06:25] -- Added extension '500' priority 2 to demo [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '500' priority 3 to demo [Mar 10 12:06:25] -- Added extension '500' priority 3 to demo [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '500' priority 4 to demo [Mar 10 12:06:25] -- Added extension '500' priority 4 to demo [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '600' priority 1 to demo [Mar 10 12:06:25] -- Added extension '600' priority 1 to demo [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '600' priority 2 to demo [Mar 10 12:06:25] -- Added extension '600' priority 2 to demo [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '600' priority 3 to demo [Mar 10 12:06:25] -- Added extension '600' priority 3 to demo [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '600' priority 4 to demo [Mar 10 12:06:25] -- Added extension '600' priority 4 to demo [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '8500' priority 1 to demo [Mar 10 12:06:25] -- Added extension '8500' priority 1 to demo [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '8500' priority 2 to demo [Mar 10 12:06:25] -- Added extension '8500' priority 2 to demo [Mar 10 12:06:25] DEBUG[18522]: pbx.c:3853 __ast_context_create: Registered context 'default' [Mar 10 12:06:25] -- Registered extension context 'default' [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 1 to default [Mar 10 12:06:25] -- Added extension 's' priority 1 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 2 to default [Mar 10 12:06:25] -- Added extension 's' priority 2 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 3 to default [Mar 10 12:06:25] -- Added extension 's' priority 3 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 4 to default [Mar 10 12:06:25] -- Added extension 's' priority 4 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 5 to default [Mar 10 12:06:25] -- Added extension 's' priority 5 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 6 to default [Mar 10 12:06:25] -- Added extension 's' priority 6 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 7 to default [Mar 10 12:06:25] -- Added extension 's' priority 7 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 8 to default [Mar 10 12:06:25] -- Added extension 's' priority 8 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 9 to default [Mar 10 12:06:25] -- Added extension 's' priority 9 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '3173241051' priority 1 to default [Mar 10 12:06:25] -- Added extension '3173241051' priority 1 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '3173241052' priority 1 to default [Mar 10 12:06:25] -- Added extension '3173241052' priority 1 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'operator' priority 1 to default [Mar 10 12:06:25] -- Added extension 'operator' priority 1 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'operator' priority 2 to default [Mar 10 12:06:25] -- Added extension 'operator' priority 2 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'operator' priority 3 to default [Mar 10 12:06:25] -- Added extension 'operator' priority 3 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'operator' priority 4 to default [Mar 10 12:06:25] -- Added extension 'operator' priority 4 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'operator' priority 5 to default [Mar 10 12:06:25] -- Added extension 'operator' priority 5 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'operator' priority 6 to default [Mar 10 12:06:25] -- Added extension 'operator' priority 6 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'operator-CHANUNAVAIL' priority 1 to default [Mar 10 12:06:25] -- Added extension 'operator-CHANUNAVAIL' priority 1 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'operator-CONGESTION' priority 1 to default [Mar 10 12:06:25] -- Added extension 'operator-CONGESTION' priority 1 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'operator-NOANSWER' priority 1 to default [Mar 10 12:06:25] -- Added extension 'operator-NOANSWER' priority 1 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'operator-BUSY' priority 1 to default [Mar 10 12:06:25] -- Added extension 'operator-BUSY' priority 1 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'operator' priority 102 to default [Mar 10 12:06:25] -- Added extension 'operator' priority 102 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '0' priority 1 to default [Mar 10 12:06:25] -- Added extension '0' priority 1 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '0' priority 2 to default [Mar 10 12:06:25] -- Added extension '0' priority 2 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '0' priority 3 to default [Mar 10 12:06:25] -- Added extension '0' priority 3 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '1' priority 1 to default [Mar 10 12:06:25] -- Added extension '1' priority 1 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '1' priority 2 to default [Mar 10 12:06:25] -- Added extension '1' priority 2 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '1' priority 3 to default [Mar 10 12:06:25] -- Added extension '1' priority 3 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '1' priority 4 to default [Mar 10 12:06:25] -- Added extension '1' priority 4 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '2' priority 1 to default [Mar 10 12:06:25] -- Added extension '2' priority 1 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '2' priority 2 to default [Mar 10 12:06:25] -- Added extension '2' priority 2 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '2' priority 3 to default [Mar 10 12:06:25] -- Added extension '2' priority 3 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '2' priority 4 to default [Mar 10 12:06:25] -- Added extension '2' priority 4 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '3' priority 1 to default [Mar 10 12:06:25] -- Added extension '3' priority 1 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '3' priority 2 to default [Mar 10 12:06:25] -- Added extension '3' priority 2 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '3' priority 3 to default [Mar 10 12:06:25] -- Added extension '3' priority 3 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '4' priority 1 to default [Mar 10 12:06:25] -- Added extension '4' priority 1 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '4' priority 2 to default [Mar 10 12:06:25] -- Added extension '4' priority 2 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '4' priority 3 to default [Mar 10 12:06:25] -- Added extension '4' priority 3 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '4' priority 4 to default [Mar 10 12:06:25] -- Added extension '4' priority 4 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '22' priority 1 to default [Mar 10 12:06:25] -- Added extension '22' priority 1 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '22' priority 2 to default [Mar 10 12:06:25] -- Added extension '22' priority 2 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '711' priority 1 to default [Mar 10 12:06:25] -- Added extension '711' priority 1 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '711' priority 2 to default [Mar 10 12:06:25] -- Added extension '711' priority 2 to default [Mar 10 12:06:25] -- Including context 'smvoice-intercom' in context 'default' [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_XXX' priority 1 to default [Mar 10 12:06:25] -- Added extension '_XXX' priority 1 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_XXX' priority 2 to default [Mar 10 12:06:25] -- Added extension '_XXX' priority 2 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_XXX' priority 3 to default [Mar 10 12:06:25] -- Added extension '_XXX' priority 3 to default [Mar 10 12:06:25] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_XXX' priority 4 to default [Mar 10 12:06:25] -- Added extension '_XXX' priority 4 to default [Mar 10 12:06:25] DEBUG[18555]: chan_sip.c:4308 sip_alloc: Allocating new SIP dialog for 2c01f4fc3cc0317e799f35d0601622e1@24.123.23.170 - REGISTER (No RTP) [Mar 10 12:06:25] DEBUG[18543]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel IAX2/614 [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_*XXX' priority 1 to default [Mar 10 12:06:27] -- Added extension '_*XXX' priority 1 to default [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_*XXX' priority 2 to default [Mar 10 12:06:27] -- Added extension '_*XXX' priority 2 to default [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_*XXX' priority 3 to default [Mar 10 12:06:27] -- Added extension '_*XXX' priority 3 to default [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '7000' priority 1 to default [Mar 10 12:06:27] -- Added extension '7000' priority 1 to default [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '7001' priority 1 to default [Mar 10 12:06:27] -- Added extension '7001' priority 1 to default [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '7002' priority 1 to default [Mar 10 12:06:27] -- Added extension '7002' priority 1 to default [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_XXXXX' priority 1 to default [Mar 10 12:06:27] -- Added extension '_XXXXX' priority 1 to default [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_XXXXX' priority 2 to default [Mar 10 12:06:27] -- Added extension '_XXXXX' priority 2 to default [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'i' priority 1 to default [Mar 10 12:06:27] -- Added extension 'i' priority 1 to default [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'i' priority 2 to default [Mar 10 12:06:27] -- Added extension 'i' priority 2 to default [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'i' priority 3 to default [Mar 10 12:06:27] -- Added extension 'i' priority 3 to default [Mar 10 12:06:27] DEBUG[18522]: pbx.c:3853 __ast_context_create: Registered context 'smvoice-local' [Mar 10 12:06:27] -- Registered extension context 'smvoice-local' [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '297' priority 1 to smvoice-local [Mar 10 12:06:27] -- Added extension '297' priority 1 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '298' priority 1 to smvoice-local [Mar 10 12:06:27] -- Added extension '298' priority 1 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '299' priority 1 to smvoice-local [Mar 10 12:06:27] -- Added extension '299' priority 1 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_1XX' priority 1 to smvoice-local [Mar 10 12:06:27] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for IAX2 - 614 [Mar 10 12:06:27] DEBUG[18527]: chan_iax2.c:9636 iax2_devicestate: Checking device state for device 614 [Mar 10 12:06:27] DEBUG[18527]: chan_iax2.c:9644 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 614? addr=-1493063488, defaddr=0 maxms=0, lastms=0 [Mar 10 12:06:27] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for IAX2/614 - state 1 (Not in use) [Mar 10 12:06:27] -- Added extension '_1XX' priority 1 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_1XX' priority 2 to smvoice-local [Mar 10 12:06:27] -- Added extension '_1XX' priority 2 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_1XX' priority 3 to smvoice-local [Mar 10 12:06:27] -- Added extension '_1XX' priority 3 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_1XX' priority 4 to smvoice-local [Mar 10 12:06:27] -- Added extension '_1XX' priority 4 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_1XX' priority 5 to smvoice-local [Mar 10 12:06:27] -- Added extension '_1XX' priority 5 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_1XX' priority 6 to smvoice-local [Mar 10 12:06:27] -- Added extension '_1XX' priority 6 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_1XX' priority 7 to smvoice-local [Mar 10 12:06:27] -- Added extension '_1XX' priority 7 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_1XX-CHANUNAVAIL' priority 1 to smvoice-local [Mar 10 12:06:27] -- Added extension '_1XX-CHANUNAVAIL' priority 1 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_1XX-CONGESTION' priority 1 to smvoice-local [Mar 10 12:06:27] -- Added extension '_1XX-CONGESTION' priority 1 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_1XX-NOANSWER' priority 1 to smvoice-local [Mar 10 12:06:27] -- Added extension '_1XX-NOANSWER' priority 1 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_1XX-BUSY' priority 1 to smvoice-local [Mar 10 12:06:27] -- Added extension '_1XX-BUSY' priority 1 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_206' priority 1 to smvoice-local [Mar 10 12:06:27] -- Added extension '_206' priority 1 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_2XX' priority 1 to smvoice-local [Mar 10 12:06:27] -- Added extension '_2XX' priority 1 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_2XX' priority 2 to smvoice-local [Mar 10 12:06:27] -- Added extension '_2XX' priority 2 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_2XX' priority 3 to smvoice-local [Mar 10 12:06:27] -- Added extension '_2XX' priority 3 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_2XX' priority 4 to smvoice-local [Mar 10 12:06:27] -- Added extension '_2XX' priority 4 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_2XX' priority 5 to smvoice-local [Mar 10 12:06:27] -- Added extension '_2XX' priority 5 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_2XX' priority 6 to smvoice-local [Mar 10 12:06:27] -- Added extension '_2XX' priority 6 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_2XX' priority 7 to smvoice-local [Mar 10 12:06:27] -- Added extension '_2XX' priority 7 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_2XX-CHANUNAVAIL' priority 1 to smvoice-local [Mar 10 12:06:27] DEBUG[18543]: db.c:197 ast_db_get: Unable to find key 'si-000fd300002e' in family 'iax/provisioning/cache' [Mar 10 12:06:27] -- Added extension '_2XX-CHANUNAVAIL' priority 1 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_2XX-CONGESTION' priority 1 to smvoice-local [Mar 10 12:06:27] -- Added extension '_2XX-CONGESTION' priority 1 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_2XX-NOANSWER' priority 1 to smvoice-local [Mar 10 12:06:27] -- Added extension '_2XX-NOANSWER' priority 1 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_2XX-BUSY' priority 1 to smvoice-local [Mar 10 12:06:27] -- Added extension '_2XX-BUSY' priority 1 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_4XX' priority 1 to smvoice-local [Mar 10 12:06:27] -- Added extension '_4XX' priority 1 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_4XX' priority 2 to smvoice-local [Mar 10 12:06:27] -- Added extension '_4XX' priority 2 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_4XX' priority 3 to smvoice-local [Mar 10 12:06:27] -- Added extension '_4XX' priority 3 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_4XX' priority 4 to smvoice-local [Mar 10 12:06:27] -- Added extension '_4XX' priority 4 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_4XX-CHANUNAVAIL' priority 1 to smvoice-local [Mar 10 12:06:27] -- Added extension '_4XX-CHANUNAVAIL' priority 1 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_4XX-CONGESTION' priority 1 to smvoice-local [Mar 10 12:06:27] -- Added extension '_4XX-CONGESTION' priority 1 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_4XX-NOANSWER' priority 1 to smvoice-local [Mar 10 12:06:27] -- Added extension '_4XX-NOANSWER' priority 1 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_4XX-BUSY' priority 1 to smvoice-local [Mar 10 12:06:27] -- Added extension '_4XX-BUSY' priority 1 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_5XX' priority 1 to smvoice-local [Mar 10 12:06:27] -- Added extension '_5XX' priority 1 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_5XX' priority 2 to smvoice-local [Mar 10 12:06:27] -- Added extension '_5XX' priority 2 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_5XX' priority 3 to smvoice-local [Mar 10 12:06:27] -- Added extension '_5XX' priority 3 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_5XX' priority 4 to smvoice-local [Mar 10 12:06:27] -- Added extension '_5XX' priority 4 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_5XX' priority 5 to smvoice-local [Mar 10 12:06:27] -- Added extension '_5XX' priority 5 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_5XX' priority 6 to smvoice-local [Mar 10 12:06:27] -- Added extension '_5XX' priority 6 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_5XX' priority 7 to smvoice-local [Mar 10 12:06:27] -- Added extension '_5XX' priority 7 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_5XX-CHANUNAVAIL' priority 1 to smvoice-local [Mar 10 12:06:27] -- Added extension '_5XX-CHANUNAVAIL' priority 1 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_5XX-CONGESTION' priority 1 to smvoice-local [Mar 10 12:06:27] -- Added extension '_5XX-CONGESTION' priority 1 to smvoice-local [Mar 10 12:06:27] DEBUG[18556]: 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. [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_5XX-NOANSWER' priority 1 to smvoice-local [Mar 10 12:06:27] -- Added extension '_5XX-NOANSWER' priority 1 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_5XX-BUSY' priority 1 to smvoice-local [Mar 10 12:06:27] -- Added extension '_5XX-BUSY' priority 1 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_6XX' priority 1 to smvoice-local [Mar 10 12:06:27] -- Added extension '_6XX' priority 1 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_6XX' priority 2 to smvoice-local [Mar 10 12:06:27] -- Added extension '_6XX' priority 2 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_6XX' priority 3 to smvoice-local [Mar 10 12:06:27] -- Added extension '_6XX' priority 3 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_6XX' priority 4 to smvoice-local [Mar 10 12:06:27] -- Added extension '_6XX' priority 4 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_6XX-CHANUNAVAIL' priority 1 to smvoice-local [Mar 10 12:06:27] -- Added extension '_6XX-CHANUNAVAIL' priority 1 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_6XX-CONGESTION' priority 1 to smvoice-local [Mar 10 12:06:27] -- Added extension '_6XX-CONGESTION' priority 1 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_6XX-NOANSWER' priority 1 to smvoice-local [Mar 10 12:06:27] -- Added extension '_6XX-NOANSWER' priority 1 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_6XX-BUSY' priority 1 to smvoice-local [Mar 10 12:06:27] -- Added extension '_6XX-BUSY' priority 1 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_8XX' priority 1 to smvoice-local [Mar 10 12:06:27] -- Added extension '_8XX' priority 1 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_8XX' priority 2 to smvoice-local [Mar 10 12:06:27] -- Added extension '_8XX' priority 2 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_8XX' priority 3 to smvoice-local [Mar 10 12:06:27] -- Added extension '_8XX' priority 3 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_8XX' priority 4 to smvoice-local [Mar 10 12:06:27] -- Added extension '_8XX' priority 4 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_8XX-CHANUNAVAIL' priority 1 to smvoice-local [Mar 10 12:06:27] -- Added extension '_8XX-CHANUNAVAIL' priority 1 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_8XX-CONGESTION' priority 1 to smvoice-local [Mar 10 12:06:27] -- Added extension '_8XX-CONGESTION' priority 1 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_8XX-NOANSWER' priority 1 to smvoice-local [Mar 10 12:06:27] -- Added extension '_8XX-NOANSWER' priority 1 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_8XX-BUSY' priority 1 to smvoice-local [Mar 10 12:06:27] -- Added extension '_8XX-BUSY' priority 1 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_*XXX' priority 1 to smvoice-local [Mar 10 12:06:27] -- Added extension '_*XXX' priority 1 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_*XXX' priority 2 to smvoice-local [Mar 10 12:06:27] -- Added extension '_*XXX' priority 2 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_*XXX' priority 3 to smvoice-local [Mar 10 12:06:27] -- Added extension '_*XXX' priority 3 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'INVALID' priority 1 to smvoice-local [Mar 10 12:06:27] -- Added extension 'INVALID' priority 1 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'INVALID' priority 2 to smvoice-local [Mar 10 12:06:27] -- Added extension 'INVALID' priority 2 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'INVALID' priority 3 to smvoice-local [Mar 10 12:06:27] -- Added extension 'INVALID' priority 3 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'i' priority 1 to smvoice-local [Mar 10 12:06:27] -- Added extension 'i' priority 1 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'i' priority 2 to smvoice-local [Mar 10 12:06:27] -- Added extension 'i' priority 2 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'i' priority 3 to smvoice-local [Mar 10 12:06:27] -- Added extension 'i' priority 3 to smvoice-local [Mar 10 12:06:27] DEBUG[18522]: pbx.c:3853 __ast_context_create: Registered context 'smvoice-sip' [Mar 10 12:06:27] -- Registered extension context 'smvoice-sip' [Mar 10 12:06:27] -- Including context 'smvoice-iaxy' in context 'smvoice-sip' [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '82473' priority 1 to smvoice-sip [Mar 10 12:06:27] -- Added extension '82473' priority 1 to smvoice-sip [Mar 10 12:06:27] DEBUG[18522]: pbx.c:3853 __ast_context_create: Registered context 'wellgate' [Mar 10 12:06:27] -- Registered extension context 'wellgate' [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 1 to wellgate [Mar 10 12:06:27] -- Added extension 's' priority 1 to wellgate [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '801' priority 1 to wellgate [Mar 10 12:06:27] -- Added extension '801' priority 1 to wellgate [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '801' priority 2 to wellgate [Mar 10 12:06:27] -- Added extension '801' priority 2 to wellgate [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '802' priority 1 to wellgate [Mar 10 12:06:27] -- Added extension '802' priority 1 to wellgate [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '802' priority 2 to wellgate [Mar 10 12:06:27] -- Added extension '802' priority 2 to wellgate [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '803' priority 1 to wellgate [Mar 10 12:06:27] -- Added extension '803' priority 1 to wellgate [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '803' priority 2 to wellgate [Mar 10 12:06:27] -- Added extension '803' priority 2 to wellgate [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '804' priority 1 to wellgate [Mar 10 12:06:27] -- Added extension '804' priority 1 to wellgate [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '804' priority 2 to wellgate [Mar 10 12:06:27] -- Added extension '804' priority 2 to wellgate [Mar 10 12:06:27] DEBUG[18522]: pbx.c:3853 __ast_context_create: Registered context 'smvoice-iaxy' [Mar 10 12:06:27] -- Registered extension context 'smvoice-iaxy' [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '50' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '50' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '50' priority 2 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '50' priority 2 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '55' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '55' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '56' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '56' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '57' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '57' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '57' priority 2 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '57' priority 2 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '57' priority 3 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '57' priority 3 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '57' priority 4 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '57' priority 4 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '58' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '58' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '58' priority 2 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '58' priority 2 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '58' priority 3 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '58' priority 3 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '58' priority 4 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '58' priority 4 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '59' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '59' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '5068012' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '5068012' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '199' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '199' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '1041' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '1041' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '1041' priority 2 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '1041' priority 2 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '1104' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '1104' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '1104' priority 2 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '1104' priority 2 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '10000' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '10000' priority 1 to smvoice-iaxy [Mar 10 12:06:27] WARNING[18522]: pbx.c:4644 add_pri: Unable to register extension '59', priority 1 in 'smvoice-iaxy', already in use [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '59' priority 2 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '59' priority 2 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '59' priority 3 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '59' priority 3 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '59' priority 4 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '59' priority 4 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '*70' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '*70' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '*71' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '*71' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '*72' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '*72' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '*73' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '*73' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '*74' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '*74' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '*75' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '*75' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '*76' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '*76' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '*77' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '*77' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '86' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '86' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '777' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '777' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '777' priority 2 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '777' priority 2 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '777' priority 3 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '777' priority 3 to smvoice-iaxy [Mar 10 12:06:27] -- Including context 'smvoice-intercom' in context 'smvoice-iaxy' [Mar 10 12:06:27] -- Including context 'smvoice-local' in context 'smvoice-iaxy' [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '7000' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '7000' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '7001' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '7001' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '7002' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '7002' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_9XXXXXXX' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_9XXXXXXX' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18543]: iax2-provision.c:253 iax_provision_version: Unable to create provisioning packet for 'si-000fd300002e' [Mar 10 12:06:27] DEBUG[18544]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel IAX2/651 [Mar 10 12:06:27] DEBUG[18544]: db.c:197 ast_db_get: Unable to find key 'si-000364000738' in family 'iax/provisioning/cache' [Mar 10 12:06:27] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for IAX2 - 651 [Mar 10 12:06:27] DEBUG[18527]: chan_iax2.c:9636 iax2_devicestate: Checking device state for device 651 [Mar 10 12:06:27] DEBUG[18527]: chan_iax2.c:9644 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 651? addr=-1459509056, defaddr=0 maxms=0, lastms=0 [Mar 10 12:06:27] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for IAX2/651 - state 1 (Not in use) [Mar 10 12:06:27] DEBUG[18544]: iax2-provision.c:253 iax_provision_version: Unable to create provisioning packet for 'si-000364000738' [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_9XXXXXXX' priority 2 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_9XXXXXXX' priority 2 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_9XXXXXXX-CHANUNAVAIL' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_9XXXXXXX-CHANUNAVAIL' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_9XXXXXXX-CHANUNAVAIL' priority 2 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_9XXXXXXX-CHANUNAVAIL' priority 2 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_9XXXXXXX-CONGESTION' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_9XXXXXXX-CONGESTION' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_9XXXXXXX-BUSY' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_9XXXXXXX-BUSY' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_9XXXXXXXXXX' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_9XXXXXXXXXX' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_9XXXXXXXXXX' priority 2 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_9XXXXXXXXXX' priority 2 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_9XXXXXXXXXX-CHANUNAVAIL' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_9XXXXXXXXXX-CHANUNAVAIL' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_9XXXXXXXXXX-CHANUNAVAIL' priority 2 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_9XXXXXXXXXX-CHANUNAVAIL' priority 2 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_9XXXXXXXXXX-CONGESTION' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_9XXXXXXXXXX-CONGESTION' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_9XXXXXXXXXX-BUSY' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_9XXXXXXXXXX-BUSY' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_91XXXXXXXXXX' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_91XXXXXXXXXX' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_91XXXXXXXXXX' priority 2 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_91XXXXXXXXXX' priority 2 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_91XXXXXXXXXX-CHANUNAVAIL' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_91XXXXXXXXXX-CHANUNAVAIL' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_91XXXXXXXXXX-CHANUNAVAIL' priority 2 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_91XXXXXXXXXX-CHANUNAVAIL' priority 2 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_91XXXXXXXXXX-CONGESTION' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_91XXXXXXXXXX-CONGESTION' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_91XXXXXXXXXX-BUSY' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_91XXXXXXXXXX-BUSY' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_8XXXXXXX' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_8XXXXXXX' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_8XXXXXXX' priority 2 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_8XXXXXXX' priority 2 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_8XXXXXXX' priority 3 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_8XXXXXXX' priority 3 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_8XXXXXXX-CHANUNAVAIL' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_8XXXXXXX-CHANUNAVAIL' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_8XXXXXXX-CHANUNAVAIL' priority 2 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_8XXXXXXX-CHANUNAVAIL' priority 2 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_8XXXXXXX-CONGESTION' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_8XXXXXXX-CONGESTION' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_8XXXXXXX-CONGESTION' priority 2 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_8XXXXXXX-CONGESTION' priority 2 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_8XXXXXXXXXX' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_8XXXXXXXXXX' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_8XXXXXXXXXX' priority 2 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_8XXXXXXXXXX' priority 2 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_8XXXXXXXXXX' priority 3 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_8XXXXXXXXXX' priority 3 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_8XXXXXXXXXX-CHANUNAVAIL' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_8XXXXXXXXXX-CHANUNAVAIL' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_8XXXXXXXXXX-CHANUNAVAIL' priority 2 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_8XXXXXXXXXX-CHANUNAVAIL' priority 2 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_8XXXXXXXXXX-CONGESTION' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_8XXXXXXXXXX-CONGESTION' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_8XXXXXXXXXX-CONGESTION' priority 2 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_8XXXXXXXXXX-CONGESTION' priority 2 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_81XXXXXXXXXX' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_81XXXXXXXXXX' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_81XXXXXXXXXX' priority 2 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_81XXXXXXXXXX' priority 2 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_81XXXXXXXXXX' priority 3 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_81XXXXXXXXXX' priority 3 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_81XXXXXXXXXX-CHANUNAVAIL' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_81XXXXXXXXXX-CHANUNAVAIL' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_81XXXXXXXXXX-CHANUNAVAIL' priority 2 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_81XXXXXXXXXX-CHANUNAVAIL' priority 2 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_81XXXXXXXXXX-CONGESTION' priority 1 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_81XXXXXXXXXX-CONGESTION' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_81XXXXXXXXXX-CONGESTION' priority 2 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_81XXXXXXXXXX-CONGESTION' priority 2 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_7XXXXXXX' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18555]: chan_sip.c:7287 transmit_register: Scheduled a registration timeout for sip.broadvoice.com id #86 [Mar 10 12:06:27] DEBUG[18557]: 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. [Mar 10 12:06:27] -- Added extension '_7XXXXXXX' priority 1 to smvoice-iaxy [Mar 10 12:06:27] REGISTER attempt 1 to 3173241052@sip.broadvoice.com [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_7XXXXXXX' priority 2 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_7XXXXXXX' priority 2 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_7XXXXXXX' priority 3 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_7XXXXXXX' priority 3 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_71XXXXXXXXXX' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18555]: chan_sip.c:15008 do_monitor: chan_sip: ast_sched_runq ran 29 all at once [Mar 10 12:06:27] -- Added extension '_71XXXXXXXXXX' priority 1 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_7XXXXXXXXXXX' priority 2 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 2c01f4fc3cc0317e799f35d0601622e1@24.123.23.170 Their Tag Our tag: as4b5dac5a [Mar 10 12:06:27] DEBUG[18555]: chan_sip.c:4308 sip_alloc: Allocating new SIP dialog for 55ae66469f7f@24.123.23.170 - REGISTER (No RTP) [Mar 10 12:06:27] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:27] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66469f7f@24.123.23.170 Their Tag ad4bac0c41ca9130ca8935d73f9d9cfb Our tag: as745f2deb [Mar 10 12:06:27] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 2c01f4fc3cc0317e799f35d0601622e1@24.123.23.170 Their Tag Our tag: as4b5dac5a [Mar 10 12:06:27] -- Added extension '_7XXXXXXXXXXX' priority 2 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18555]: chan_sip.c:4308 sip_alloc: Allocating new SIP dialog for 1319586744@192.168.1.97 - REGISTER (No RTP) [Mar 10 12:06:27] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:27] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 1319586744@192.168.1.97 Their Tag 4210231806 Our tag: as609501e8 [Mar 10 12:06:27] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66469f7f@24.123.23.170 Their Tag ad4bac0c41ca9130ca8935d73f9d9cfb Our tag: as745f2deb [Mar 10 12:06:27] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 2c01f4fc3cc0317e799f35d0601622e1@24.123.23.170 Their Tag Our tag: as4b5dac5a [Mar 10 12:06:27] DEBUG[18555]: chan_sip.c:4308 sip_alloc: Allocating new SIP dialog for 55ae664696c0@24.123.23.170 - REGISTER (No RTP) [Mar 10 12:06:27] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:27] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae664696c0@24.123.23.170 Their Tag 93cc73e47e62b2ad602ae1a53505884b Our tag: as7d71073d [Mar 10 12:06:27] DEBUG[18555]: chan_sip.c:4359 find_call: = Found Their Call ID: 1319586744@192.168.1.97 Their Tag 4210231806 Our tag: as609501e8 [Mar 10 12:06:27] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:27] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 60, ours 60) [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '_71XXXXXXXXXX' priority 3 to smvoice-iaxy [Mar 10 12:06:27] -- Added extension '_71XXXXXXXXXX' priority 3 to smvoice-iaxy [Mar 10 12:06:27] DEBUG[18522]: pbx.c:3853 __ast_context_create: Registered context 'smvoice-return-voicemail' [Mar 10 12:06:27] -- Registered extension context 'smvoice-return-voicemail' [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 1 to smvoice-return-voicemail [Mar 10 12:06:27] -- Added extension 's' priority 1 to smvoice-return-voicemail [Mar 10 12:06:27] DEBUG[18522]: pbx.c:3853 __ast_context_create: Registered context 'smvoice-voicemail' [Mar 10 12:06:27] -- Registered extension context 'smvoice-voicemail' [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '0' priority 1 to smvoice-voicemail [Mar 10 12:06:27] -- Added extension '0' priority 1 to smvoice-voicemail [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '1' priority 1 to smvoice-voicemail [Mar 10 12:06:27] -- Added extension '1' priority 1 to smvoice-voicemail [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '2' priority 1 to smvoice-voicemail [Mar 10 12:06:27] -- Added extension '2' priority 1 to smvoice-voicemail [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '3' priority 1 to smvoice-voicemail [Mar 10 12:06:27] -- Added extension '3' priority 1 to smvoice-voicemail [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '101' priority 1 to smvoice-voicemail [Mar 10 12:06:27] -- Added extension '101' priority 1 to smvoice-voicemail [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '510' priority 1 to smvoice-voicemail [Mar 10 12:06:27] -- Added extension '510' priority 1 to smvoice-voicemail [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '511' priority 1 to smvoice-voicemail [Mar 10 12:06:27] -- Added extension '511' priority 1 to smvoice-voicemail [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '512' priority 1 to smvoice-voicemail [Mar 10 12:06:27] -- Added extension '512' priority 1 to smvoice-voicemail [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '513' priority 1 to smvoice-voicemail [Mar 10 12:06:27] -- Added extension '513' priority 1 to smvoice-voicemail [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '514' priority 1 to smvoice-voicemail [Mar 10 12:06:27] -- Added extension '514' priority 1 to smvoice-voicemail [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '515' priority 1 to smvoice-voicemail [Mar 10 12:06:27] -- Added extension '515' priority 1 to smvoice-voicemail [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '806' priority 1 to smvoice-voicemail [Mar 10 12:06:27] -- Added extension '806' priority 1 to smvoice-voicemail [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '205' priority 1 to smvoice-voicemail [Mar 10 12:06:27] -- Added extension '205' priority 1 to smvoice-voicemail [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '522' priority 1 to smvoice-voicemail [Mar 10 12:06:27] -- Added extension '522' priority 1 to smvoice-voicemail [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '605' priority 1 to smvoice-voicemail [Mar 10 12:06:27] -- Added extension '605' priority 1 to smvoice-voicemail [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '801' priority 1 to smvoice-voicemail [Mar 10 12:06:27] -- Added extension '801' priority 1 to smvoice-voicemail [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '210' priority 1 to smvoice-voicemail [Mar 10 12:06:27] -- Added extension '210' priority 1 to smvoice-voicemail [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '216' priority 1 to smvoice-voicemail [Mar 10 12:06:27] -- Added extension '216' priority 1 to smvoice-voicemail [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '592' priority 1 to smvoice-voicemail [Mar 10 12:06:27] -- Added extension '592' priority 1 to smvoice-voicemail [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '2134' priority 1 to smvoice-voicemail [Mar 10 12:06:27] -- Added extension '2134' priority 1 to smvoice-voicemail [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '209' priority 1 to smvoice-voicemail [Mar 10 12:06:27] -- Added extension '209' priority 1 to smvoice-voicemail [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '204' priority 1 to smvoice-voicemail [Mar 10 12:06:27] -- Added extension '204' priority 1 to smvoice-voicemail [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '401' priority 1 to smvoice-voicemail [Mar 10 12:06:27] -- Added extension '401' priority 1 to smvoice-voicemail [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '402' priority 1 to smvoice-voicemail [Mar 10 12:06:27] -- Added extension '402' priority 1 to smvoice-voicemail [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '403' priority 1 to smvoice-voicemail [Mar 10 12:06:27] -- Added extension '403' priority 1 to smvoice-voicemail [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '404' priority 1 to smvoice-voicemail [Mar 10 12:06:27] -- Added extension '404' priority 1 to smvoice-voicemail [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '528' priority 1 to smvoice-voicemail [Mar 10 12:06:27] -- Added extension '528' priority 1 to smvoice-voicemail [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '530' priority 1 to smvoice-voicemail [Mar 10 12:06:27] -- Added extension '530' priority 1 to smvoice-voicemail [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '531' priority 1 to smvoice-voicemail [Mar 10 12:06:27] -- Added extension '531' priority 1 to smvoice-voicemail [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '606' priority 1 to smvoice-voicemail [Mar 10 12:06:27] -- Added extension '606' priority 1 to smvoice-voicemail [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '616' priority 1 to smvoice-voicemail [Mar 10 12:06:27] -- Added extension '616' priority 1 to smvoice-voicemail [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '800' priority 1 to smvoice-voicemail [Mar 10 12:06:27] -- Added extension '800' priority 1 to smvoice-voicemail [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'i' priority 1 to smvoice-voicemail [Mar 10 12:06:27] -- Added extension 'i' priority 1 to smvoice-voicemail [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'i' priority 2 to smvoice-voicemail [Mar 10 12:06:27] -- Added extension 'i' priority 2 to smvoice-voicemail [Mar 10 12:06:27] DEBUG[18522]: pbx.c:3853 __ast_context_create: Registered context 'smvoice-intercom' [Mar 10 12:06:27] -- Registered extension context 'smvoice-intercom' [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '87' priority 1 to smvoice-intercom [Mar 10 12:06:27] -- Added extension '87' priority 1 to smvoice-intercom [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension '87' priority 2 to smvoice-intercom [Mar 10 12:06:27] -- Added extension '87' priority 2 to smvoice-intercom [Mar 10 12:06:27] DEBUG[18522]: pbx.c:3853 __ast_context_create: Registered context 'smvoice-incoming' [Mar 10 12:06:27] -- Registered extension context 'smvoice-incoming' [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 1 to smvoice-incoming [Mar 10 12:06:27] -- Added extension 's' priority 1 to smvoice-incoming [Mar 10 12:06:27] -- Including context 'smvoice-intercom' in context 'smvoice-incoming' [Mar 10 12:06:27] -- Including context 'smvoice-transfers' in context 'smvoice-incoming' [Mar 10 12:06:27] DEBUG[18522]: pbx.c:3853 __ast_context_create: Registered context 'smvoice-incoming-kx1232' [Mar 10 12:06:27] -- Registered extension context 'smvoice-incoming-kx1232' [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 1 to smvoice-incoming-kx1232 [Mar 10 12:06:27] -- Added extension 's' priority 1 to smvoice-incoming-kx1232 [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 2 to smvoice-incoming-kx1232 [Mar 10 12:06:27] -- Added extension 's' priority 2 to smvoice-incoming-kx1232 [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 3 to smvoice-incoming-kx1232 [Mar 10 12:06:27] -- Added extension 's' priority 3 to smvoice-incoming-kx1232 [Mar 10 12:06:27] DEBUG[18522]: pbx.c:3853 __ast_context_create: Registered context 'smvoice-transfers' [Mar 10 12:06:27] -- Registered extension context 'smvoice-transfers' [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'smvoice_dial' priority 1 to smvoice-transfers [Mar 10 12:06:27] -- Added extension 'smvoice_dial' priority 1 to smvoice-transfers [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'smvoice_dial' priority 2 to smvoice-transfers [Mar 10 12:06:27] -- Added extension 'smvoice_dial' priority 2 to smvoice-transfers [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'smvoice_dial-CHANUNAVAIL' priority 1 to smvoice-transfers [Mar 10 12:06:27] -- Added extension 'smvoice_dial-CHANUNAVAIL' priority 1 to smvoice-transfers [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'smvoice_dial-CHANUNAVAIL' priority 2 to smvoice-transfers [Mar 10 12:06:27] -- Added extension 'smvoice_dial-CHANUNAVAIL' priority 2 to smvoice-transfers [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'smvoice_dial-CONGESTION' priority 1 to smvoice-transfers [Mar 10 12:06:27] -- Added extension 'smvoice_dial-CONGESTION' priority 1 to smvoice-transfers [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'smvoice_dial-CONGESTION' priority 2 to smvoice-transfers [Mar 10 12:06:27] -- Added extension 'smvoice_dial-CONGESTION' priority 2 to smvoice-transfers [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'smvoice_dial-NOANSWER' priority 1 to smvoice-transfers [Mar 10 12:06:27] -- Added extension 'smvoice_dial-NOANSWER' priority 1 to smvoice-transfers [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'smvoice_dial-NOANSWER' priority 2 to smvoice-transfers [Mar 10 12:06:27] -- Added extension 'smvoice_dial-NOANSWER' priority 2 to smvoice-transfers [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'smvoice_dial-BUSY' priority 1 to smvoice-transfers [Mar 10 12:06:27] -- Added extension 'smvoice_dial-BUSY' priority 1 to smvoice-transfers [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'smvoice_dial-BUSY' priority 2 to smvoice-transfers [Mar 10 12:06:27] -- Added extension 'smvoice_dial-BUSY' priority 2 to smvoice-transfers [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'smvoice_dial_no_extension' priority 1 to smvoice-transfers [Mar 10 12:06:27] -- Added extension 'smvoice_dial_no_extension' priority 1 to smvoice-transfers [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'smvoice_dial_no_extension' priority 2 to smvoice-transfers [Mar 10 12:06:27] -- Added extension 'smvoice_dial_no_extension' priority 2 to smvoice-transfers [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'smvoice_dial_no_extension-CHANUNAVAIL' priority 1 to smvoice-transfers [Mar 10 12:06:27] -- Added extension 'smvoice_dial_no_extension-CHANUNAVAIL' priority 1 to smvoice-transfers [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'smvoice_dial_no_extension-CHANUNAVAIL' priority 2 to smvoice-transfers [Mar 10 12:06:27] -- Added extension 'smvoice_dial_no_extension-CHANUNAVAIL' priority 2 to smvoice-transfers [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'smvoice_dial_no_extension-CONGESTION' priority 1 to smvoice-transfers [Mar 10 12:06:27] -- Added extension 'smvoice_dial_no_extension-CONGESTION' priority 1 to smvoice-transfers [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'smvoice_dial_no_extension-BUSY' priority 1 to smvoice-transfers [Mar 10 12:06:27] -- Added extension 'smvoice_dial_no_extension-BUSY' priority 1 to smvoice-transfers [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'smvoice_dial_no_extension-NOANSWER' priority 1 to smvoice-transfers [Mar 10 12:06:27] -- Added extension 'smvoice_dial_no_extension-NOANSWER' priority 1 to smvoice-transfers [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'smvoice_dial_no_extension-NOANSWER' priority 2 to smvoice-transfers [Mar 10 12:06:27] -- Added extension 'smvoice_dial_no_extension-NOANSWER' priority 2 to smvoice-transfers [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'smvoice_dial_goto_voicemail' priority 1 to smvoice-transfers [Mar 10 12:06:27] -- Added extension 'smvoice_dial_goto_voicemail' priority 1 to smvoice-transfers [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'smvoice_dial_goto_voicemail' priority 2 to smvoice-transfers [Mar 10 12:06:27] -- Added extension 'smvoice_dial_goto_voicemail' priority 2 to smvoice-transfers [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'smvoice_dial_goto_voicemail-CHANUNAVAIL' priority 1 to smvoice-transfers [Mar 10 12:06:27] -- Added extension 'smvoice_dial_goto_voicemail-CHANUNAVAIL' priority 1 to smvoice-transfers [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'smvoice_dial_goto_voicemail-CONGESTION' priority 1 to smvoice-transfers [Mar 10 12:06:27] -- Added extension 'smvoice_dial_goto_voicemail-CONGESTION' priority 1 to smvoice-transfers [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'smvoice_dial_goto_voicemail-NOANSWER' priority 1 to smvoice-transfers [Mar 10 12:06:27] -- Added extension 'smvoice_dial_goto_voicemail-NOANSWER' priority 1 to smvoice-transfers [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'smvoice_dial_goto_voicemail-BUSY' priority 1 to smvoice-transfers [Mar 10 12:06:27] -- Added extension 'smvoice_dial_goto_voicemail-BUSY' priority 1 to smvoice-transfers [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'smvoice_conference' priority 1 to smvoice-transfers [Mar 10 12:06:27] -- Added extension 'smvoice_conference' priority 1 to smvoice-transfers [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'smvoice_sendtext' priority 1 to smvoice-transfers [Mar 10 12:06:27] -- Added extension 'smvoice_sendtext' priority 1 to smvoice-transfers [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'smvoice_sendtext' priority 2 to smvoice-transfers [Mar 10 12:06:27] -- Added extension 'smvoice_sendtext' priority 2 to smvoice-transfers [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'smvoice_sendtext' priority 3 to smvoice-transfers [Mar 10 12:06:27] -- Added extension 'smvoice_sendtext' priority 3 to smvoice-transfers [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'smvoice_no_callprogress' priority 1 to smvoice-transfers [Mar 10 12:06:27] -- Added extension 'smvoice_no_callprogress' priority 1 to smvoice-transfers [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'smvoice_callprogress' priority 1 to smvoice-transfers [Mar 10 12:06:27] -- Added extension 'smvoice_callprogress' priority 1 to smvoice-transfers [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'smvoice_callprogress' priority 2 to smvoice-transfers [Mar 10 12:06:27] -- Added extension 'smvoice_callprogress' priority 2 to smvoice-transfers [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'smvoice_callprogress' priority 3 to smvoice-transfers [Mar 10 12:06:27] -- Added extension 'smvoice_callprogress' priority 3 to smvoice-transfers [Mar 10 12:06:27] DEBUG[18522]: pbx.c:3853 __ast_context_create: Registered context 'smvoice-homework-hotline' [Mar 10 12:06:27] -- Registered extension context 'smvoice-homework-hotline' [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 1 to smvoice-homework-hotline [Mar 10 12:06:27] -- Added extension 's' priority 1 to smvoice-homework-hotline [Mar 10 12:06:27] DEBUG[18522]: pbx.c:3853 __ast_context_create: Registered context 'smvoice-faxout' [Mar 10 12:06:27] -- Registered extension context 'smvoice-faxout' [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'smvoice_faxout' priority 1 to smvoice-faxout [Mar 10 12:06:27] -- Added extension 'smvoice_faxout' priority 1 to smvoice-faxout [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'smvoice_faxout' priority 2 to smvoice-faxout [Mar 10 12:06:27] -- Added extension 'smvoice_faxout' priority 2 to smvoice-faxout [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'smvoice_faxout' priority 3 to smvoice-faxout [Mar 10 12:06:27] -- Added extension 'smvoice_faxout' priority 3 to smvoice-faxout [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'smvoice_faxout' priority 4 to smvoice-faxout [Mar 10 12:06:27] -- Added extension 'smvoice_faxout' priority 4 to smvoice-faxout [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'smvoice_faxout' priority 104 to smvoice-faxout [Mar 10 12:06:27] -- Added extension 'smvoice_faxout' priority 104 to smvoice-faxout [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'smvoice_faxout' priority 105 to smvoice-faxout [Mar 10 12:06:27] -- Added extension 'smvoice_faxout' priority 105 to smvoice-faxout [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'i' priority 1 to smvoice-faxout [Mar 10 12:06:27] -- Added extension 'i' priority 1 to smvoice-faxout [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'i' priority 2 to smvoice-faxout [Mar 10 12:06:27] -- Added extension 'i' priority 2 to smvoice-faxout [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 't' priority 1 to smvoice-faxout [Mar 10 12:06:27] -- Added extension 't' priority 1 to smvoice-faxout [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 't' priority 2 to smvoice-faxout [Mar 10 12:06:27] -- Added extension 't' priority 2 to smvoice-faxout [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'T' priority 1 to smvoice-faxout [Mar 10 12:06:27] -- Added extension 'T' priority 1 to smvoice-faxout [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'T' priority 2 to smvoice-faxout [Mar 10 12:06:27] -- Added extension 'T' priority 2 to smvoice-faxout [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'failed' priority 1 to smvoice-faxout [Mar 10 12:06:27] -- Added extension 'failed' priority 1 to smvoice-faxout [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'failed' priority 2 to smvoice-faxout [Mar 10 12:06:27] -- Added extension 'failed' priority 2 to smvoice-faxout [Mar 10 12:06:27] DEBUG[18522]: pbx.c:3853 __ast_context_create: Registered context 'smvoice-dialout' [Mar 10 12:06:27] -- Registered extension context 'smvoice-dialout' [Mar 10 12:06:27] -- Including context 'smvoice-intercom' in context 'smvoice-dialout' [Mar 10 12:06:27] -- Including context 'smvoice-transfers' in context 'smvoice-dialout' [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'i' priority 1 to smvoice-dialout [Mar 10 12:06:27] -- Added extension 'i' priority 1 to smvoice-dialout [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 't' priority 1 to smvoice-dialout [Mar 10 12:06:27] -- Added extension 't' priority 1 to smvoice-dialout [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'T' priority 1 to smvoice-dialout [Mar 10 12:06:27] -- Added extension 'T' priority 1 to smvoice-dialout [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'failed' priority 1 to smvoice-dialout [Mar 10 12:06:27] -- Added extension 'failed' priority 1 to smvoice-dialout [Mar 10 12:06:27] DEBUG[18522]: pbx.c:3853 __ast_context_create: Registered context 'iax_devcentos64_to_unifiedpaging' [Mar 10 12:06:27] -- Registered extension context 'iax_devcentos64_to_unifiedpaging' [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 's' priority 1 to iax_devcentos64_to_unifiedpaging [Mar 10 12:06:27] -- Added extension 's' priority 1 to iax_devcentos64_to_unifiedpaging [Mar 10 12:06:27] DEBUG[18522]: pbx.c:3853 __ast_context_create: Registered context 'smvoice-testing' [Mar 10 12:06:27] -- Registered extension context 'smvoice-testing' [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'call_cell' priority 1 to smvoice-testing [Mar 10 12:06:27] -- Added extension 'call_cell' priority 1 to smvoice-testing [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'call_cell' priority 2 to smvoice-testing [Mar 10 12:06:27] -- Added extension 'call_cell' priority 2 to smvoice-testing [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'call_cell' priority 3 to smvoice-testing [Mar 10 12:06:27] -- Added extension 'call_cell' priority 3 to smvoice-testing [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'call_cell' priority 4 to smvoice-testing [Mar 10 12:06:27] -- Added extension 'call_cell' priority 4 to smvoice-testing [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'failed' priority 1 to smvoice-testing [Mar 10 12:06:27] -- Added extension 'failed' priority 1 to smvoice-testing [Mar 10 12:06:27] DEBUG[18522]: pbx.c:4822 ast_add_extension2: Added extension 'failed' priority 2 to smvoice-testing [Mar 10 12:06:27] -- Added extension 'failed' priority 2 to smvoice-testing [Mar 10 12:06:27] DEBUG[18522]: pbx.c:3928 ast_merge_contexts_and_delete: must remove any reg pbx_config [Mar 10 12:06:27] DEBUG[18522]: pbx.c:5265 __ast_context_destroy: check ctx ael-demo pbx_ael [Mar 10 12:06:27] DEBUG[18522]: pbx.c:5265 __ast_context_destroy: check ctx macro-std-exten-ael pbx_ael [Mar 10 12:06:27] DEBUG[18522]: pbx.c:5265 __ast_context_destroy: check ctx parkedcalls res_features [Mar 10 12:06:27] pbx_config.so => (Text Extension Configuration) [Mar 10 12:06:27] == Registered file format wav49, extension(s) WAV|wav49 [Mar 10 12:06:30] format_wav_gsm.so => (Microsoft WAV format (Proprietary GSM)) [Mar 10 12:06:30] == Registered application 'LookupCIDName' [Mar 10 12:06:30] app_lookupcidname.so => (Look up CallerID Name from local database) [Mar 10 12:06:30] == Registered application 'Directory' [Mar 10 12:06:30] app_directory.so => (Extension Directory) [Mar 10 12:06:30] == Registered application 'Flash' [Mar 10 12:06:30] app_flash.so => (Flash channel application) [Mar 10 12:06:28] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae664696c0@24.123.23.170 Their Tag 93cc73e47e62b2ad602ae1a53505884b Our tag: as7d71073d [Mar 10 12:06:30] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 1319586744@192.168.1.97 Their Tag 4210231806 Our tag: as609501e8 [Mar 10 12:06:30] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66469f7f@24.123.23.170 Their Tag ad4bac0c41ca9130ca8935d73f9d9cfb Our tag: as745f2deb [Mar 10 12:06:30] DEBUG[18555]: chan_sip.c:4359 find_call: = Found Their Call ID: 2c01f4fc3cc0317e799f35d0601622e1@24.123.23.170 Their Tag Our tag: as4b5dac5a [Mar 10 12:06:30] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '2c01f4fc3cc0317e799f35d0601622e1@24.123.23.170' of Request 102: Match Not Found [Mar 10 12:06:30] == Registered application 'ZapScan' [Mar 10 12:06:30] app_zapscan.so => (Scan Zap channels application) [Mar 10 12:06:30] DEBUG[18555]: chan_sip.c:1619 initialize_initreq: Initializing already initialized SIP dialog 2c01f4fc3cc0317e799f35d0601622e1@24.123.23.170 (presumably reinvite) [Mar 10 12:06:30] REGISTER attempt 2 to 3173241052@sip.broadvoice.com [Mar 10 12:06:30] DEBUG[18555]: chan_sip.c:4308 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Mar 10 12:06:30] == Registered file format iLBC, extension(s) ilbc [Mar 10 12:06:30] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 30a8ac460b4ce7b1044a8f404a92e654@24.123.23.170 Their Tag Our tag: as631ff9e7 [Mar 10 12:06:30] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae664696c0@24.123.23.170 Their Tag 93cc73e47e62b2ad602ae1a53505884b Our tag: as7d71073d [Mar 10 12:06:30] DEBUG[18555]: chan_sip.c:4359 find_call: = Found Their Call ID: 1319586744@192.168.1.97 Their Tag 4210231806 Our tag: as609501e8 [Mar 10 12:06:30] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:30] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 60, ours 60) [Mar 10 12:06:30] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 30a8ac460b4ce7b1044a8f404a92e654@24.123.23.170 Their Tag Our tag: as631ff9e7 [Mar 10 12:06:30] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae664696c0@24.123.23.170 Their Tag 93cc73e47e62b2ad602ae1a53505884b Our tag: as7d71073d [Mar 10 12:06:30] DEBUG[18555]: chan_sip.c:4359 find_call: = Found Their Call ID: 1319586744@192.168.1.97 Their Tag 4210231806 Our tag: as609501e8 [Mar 10 12:06:30] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:28] DEBUG[18547]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel IAX2/606 [Mar 10 12:06:30] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for IAX2 - 606 [Mar 10 12:06:30] DEBUG[18527]: chan_iax2.c:9636 iax2_devicestate: Checking device state for device 606 [Mar 10 12:06:30] DEBUG[18527]: chan_iax2.c:9644 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 606? addr=-1509840704, defaddr=0 maxms=0, lastms=0 [Mar 10 12:06:30] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for IAX2/606 - state 1 (Not in use) [Mar 10 12:06:30] format_ilbc.so => (Raw iLBC data) [Mar 10 12:06:30] DEBUG[18558]: 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. [Mar 10 12:06:30] DEBUG[18522]: channel.c:524 ast_channel_register: Registered handler for 'Agent' (Call Agent Proxy Channel) [Mar 10 12:06:30] == Registered channel type 'Agent' (Call Agent Proxy Channel) [Mar 10 12:06:30] == Parsing '/etc/asterisk/agents.conf': [Mar 10 12:06:30] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/agents.conf [Mar 10 12:06:30] Found [Mar 10 12:06:30] == Registered application 'AgentLogin' [Mar 10 12:06:30] == Registered application 'AgentCallbackLogin' [Mar 10 12:06:30] == Registered application 'AgentMonitorOutgoing' [Mar 10 12:06:30] == Manager registered action Agents [Mar 10 12:06:30] == Manager registered action AgentLogoff [Mar 10 12:06:30] == Manager registered action AgentCallbackLogin [Mar 10 12:06:30] == Registered custom function AGENT [Mar 10 12:06:30] chan_agent.so => (Agent Proxy Channel) [Mar 10 12:06:30] == Registered application 'SetCallerPres' [Mar 10 12:06:30] == Registered application 'SetCallerID' [Mar 10 12:06:30] app_setcallerid.so => (Set CallerID Application) [Mar 10 12:06:30] == Registered application 'Transfer' [Mar 10 12:06:30] app_transfer.so => (Transfer) [Mar 10 12:06:30] DEBUG[18522]: channel.c:524 ast_channel_register: Registered handler for 'Local' (Local Proxy Channel Driver) [Mar 10 12:06:30] == Registered channel type 'Local' (Local Proxy Channel Driver) [Mar 10 12:06:30] chan_local.so => (Local Proxy Channel) [Mar 10 12:06:30] == Parsing '/etc/asterisk/codecs.conf': [Mar 10 12:06:30] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/codecs.conf [Mar 10 12:06:30] Found [Mar 10 12:06:30] -- codec_g726: using generic PLC [Mar 10 12:06:30] WARNING[18522]: translate.c:675 __ast_register_translator: plc_samples 160 format 6 [Mar 10 12:06:30] == Registered translator 'g726tolin' from format g726 to slin, cost 1 [Mar 10 12:06:30] DEBUG[18522]: translate.c:425 rebuild_matrix: Resetting translation matrix [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from g723 to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18547]: db.c:197 ast_db_get: Unable to find key 'si-000fd3000028' in family 'iax/provisioning/cache' [Mar 10 12:06:30] DEBUG[18547]: iax2-provision.c:253 iax_provision_version: Unable to create provisioning packet for 'si-000fd3000028' [Mar 10 12:06:30] DEBUG[18548]: chan_iax2.c:6538 socket_process: Packet arrived out of order (expecting 1, got 0) (frametype = 6, subclass = 13) [Mar 10 12:06:30] DEBUG[18548]: chan_iax2.c:6545 socket_process: Acking anyway [Mar 10 12:06:30] -- Saved useragent "UTSTARCOM F1000/Device ID-0007ba26174b" for peer 551 [Mar 10 12:06:30] DEBUG[18555]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/551 [Mar 10 12:06:30] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 551 [Mar 10 12:06:30] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 551 [Mar 10 12:06:30] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/551 - state 1 (Not in use) [Mar 10 12:06:30] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 30a8ac460b4ce7b1044a8f404a92e654@24.123.23.170 Their Tag Our tag: as631ff9e7 [Mar 10 12:06:30] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae664696c0@24.123.23.170 Their Tag 93cc73e47e62b2ad602ae1a53505884b Our tag: as7d71073d [Mar 10 12:06:30] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 1319586744@192.168.1.97 Their Tag 4210231806 Our tag: as609501e8 [Mar 10 12:06:30] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66469f7f@24.123.23.170 Their Tag ad4bac0c41ca9130ca8935d73f9d9cfb Our tag: as745f2deb [Mar 10 12:06:30] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 2c01f4fc3cc0317e799f35d0601622e1@24.123.23.170 Their Tag Our tag: as005b8eb2 [Mar 10 12:06:30] DEBUG[18555]: chan_sip.c:4308 sip_alloc: Allocating new SIP dialog for 55ae66476793@24.123.23.170 - REGISTER (No RTP) [Mar 10 12:06:30] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:30] == Registered translator 'lintog726' from format slin to g726, cost 3 [Mar 10 12:06:30] DEBUG[18522]: translate.c:425 rebuild_matrix: Resetting translation matrix [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from g723 to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] WARNING[18522]: translate.c:675 __ast_register_translator: plc_samples 160 format 6 [Mar 10 12:06:30] == Registered translator 'g726aal2tolin' from format g726aal2 to slin, cost 1 [Mar 10 12:06:30] DEBUG[18522]: translate.c:425 rebuild_matrix: Resetting translation matrix [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from g723 to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from ulaw to g723, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from ulaw to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from ulaw to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from ulaw to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from ulaw to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18555]: chan_sip.c:4359 find_call: = Found Their Call ID: 55ae66476793@24.123.23.170 Their Tag 8008f64acdedd308808ed20d8e710732 Our tag: as28e6a2b1 [Mar 10 12:06:30] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:30] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 39007, ours 39007) [Mar 10 12:06:30] DEBUG[18559]: app_queue.c:546 changethread: Device 'SIP/551' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] == Registered translator 'lintog726aal2' from format slin to g726aal2, cost 1 [Mar 10 12:06:30] DEBUG[18522]: translate.c:425 rebuild_matrix: Resetting translation matrix [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to ulaw, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from g723 to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to ulaw, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from ulaw to g723, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from ulaw to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from ulaw to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from ulaw to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from ulaw to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to ulaw, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] == Registered translator 'g726aal2tog726' from format g726aal2 to g726, cost 1 [Mar 10 12:06:30] DEBUG[18522]: translate.c:425 rebuild_matrix: Resetting translation matrix [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to ulaw, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from g723 to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to ulaw, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from ulaw to g723, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from ulaw to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from ulaw to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from ulaw to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 4 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6 [Mar 10 12:06:30] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to ulaw, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from g723 to unknown, via 4 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 4 [Mar 10 12:06:30] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66476793@24.123.23.170 Their Tag 8008f64acdedd308808ed20d8e710732 Our tag: as28e6a2b1 [Mar 10 12:06:32] DEBUG[18555]: chan_sip.c:4359 find_call: = Found Their Call ID: 30a8ac460b4ce7b1044a8f404a92e654@24.123.23.170 Their Tag Our tag: as631ff9e7 [Mar 10 12:06:32] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '30a8ac460b4ce7b1044a8f404a92e654@24.123.23.170' of Request 102: Match Not Found [Mar 10 12:06:32] NOTICE[18555]: chan_sip.c:12120 handle_response_peerpoke: Peer 'Broadvoice' is now Lagged. (2674ms / 2000ms) [Mar 10 12:06:32] DEBUG[18555]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/Broadvoice [Mar 10 12:06:32] DEBUG[18555]: chan_sip.c:1878 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #92)) [Mar 10 12:06:32] == Registered translator 'g726tog726aal2' from format g726 to g726aal2, cost 1 [Mar 10 12:06:30] DEBUG[18552]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel IAX2/616 [Mar 10 12:06:32] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - Broadvoice [Mar 10 12:06:32] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer Broadvoice [Mar 10 12:06:32] DEBUG[18522]: translate.c:425 rebuild_matrix: Resetting translation matrix [Mar 10 12:06:32] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/Broadvoice - state 5 (Unavailable) [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to ulaw, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from g723 to unknown, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to ulaw, via 6 [Mar 10 12:06:32] DEBUG[18552]: db.c:197 ast_db_get: Unable to find key 'si-000fd3000124' in family 'iax/provisioning/cache' [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from ulaw to g723, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from ulaw to unknown, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from ulaw to unknown, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from ulaw to unknown, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 4 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 4 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 4 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 4 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 4 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:32] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for IAX2 - 616 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Mar 10 12:06:32] DEBUG[18527]: chan_iax2.c:9636 iax2_devicestate: Checking device state for device 616 [Mar 10 12:06:32] DEBUG[18527]: chan_iax2.c:9644 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 616? addr=1159759178, defaddr=0 maxms=0, lastms=0 [Mar 10 12:06:32] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for IAX2/616 - state 1 (Not in use) [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from g723 to unknown, via 4 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 4 [Mar 10 12:06:32] codec_g726.so => (ITU G.726-32kbps G726 Transcoder) [Mar 10 12:06:32] DEBUG[18555]: sched.c:204 sched_settime: Request to schedule in the past?!?! [Mar 10 12:06:32] DEBUG[18555]: chan_sip.c:1878 retrans_pkt: ** SIP timers: Rescheduling retransmission 3 to 2000 ms (t1 500 ms (Retrans id #92)) [Mar 10 12:06:32] Really destroying SIP dialog '30a8ac460b4ce7b1044a8f404a92e654@24.123.23.170' Method: OPTIONS [Mar 10 12:06:32] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66476793@24.123.23.170 Their Tag 8008f64acdedd308808ed20d8e710732 Our tag: as28e6a2b1 [Mar 10 12:06:32] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae664696c0@24.123.23.170 Their Tag 93cc73e47e62b2ad602ae1a53505884b Our tag: as7d71073d [Mar 10 12:06:32] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 1319586744@192.168.1.97 Their Tag 4210231806 Our tag: as609501e8 [Mar 10 12:06:32] == Registered application 'ZapSendKeypadFacility' [Mar 10 12:06:32] == Parsing '/etc/asterisk/zapata.conf': [Mar 10 12:06:32] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/zapata.conf [Mar 10 12:06:32] Found [Mar 10 12:06:32] DEBUG[18560]: app_queue.c:546 changethread: Device 'SIP/Broadvoice' changed to state '5' (Unavailable) but we don't care because they're not a member of any queue. [Mar 10 12:06:32] DEBUG[18522]: chan_zap.c:1402 update_conf: Updated conferencing on 1, with 0 conference users [Mar 10 12:06:32] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66469f7f@24.123.23.170 Their Tag ad4bac0c41ca9130ca8935d73f9d9cfb Our tag: as745f2deb [Mar 10 12:06:32] DEBUG[18555]: chan_sip.c:4359 find_call: = Found Their Call ID: 2c01f4fc3cc0317e799f35d0601622e1@24.123.23.170 Their Tag Our tag: as005b8eb2 [Mar 10 12:06:32] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '2c01f4fc3cc0317e799f35d0601622e1@24.123.23.170' of Request 103: Match Not Found [Mar 10 12:06:32] DEBUG[18555]: chan_sip.c:12025 handle_response_register: Registration successful [Mar 10 12:06:32] DEBUG[18561]: 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. [Mar 10 12:06:32] -- Registered channel 1, FXS Kewlstart signalling [Mar 10 12:06:32] DEBUG[18555]: chan_sip.c:12028 handle_response_register: Cancelling timeout 86 [Mar 10 12:06:32] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66476793@24.123.23.170 Their Tag 8008f64acdedd308808ed20d8e710732 Our tag: as28e6a2b1 [Mar 10 12:06:32] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae664696c0@24.123.23.170 Their Tag 93cc73e47e62b2ad602ae1a53505884b Our tag: as7d71073d [Mar 10 12:06:32] DEBUG[18555]: chan_sip.c:4359 find_call: = Found Their Call ID: 1319586744@192.168.1.97 Their Tag 4210231806 Our tag: as609501e8 [Mar 10 12:06:32] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:32] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 61, ours 61) [Mar 10 12:06:32] DEBUG[18522]: chan_zap.c:1402 update_conf: Updated conferencing on 2, with 0 conference users [Mar 10 12:06:32] -- Registered channel 2, FXS Kewlstart signalling [Mar 10 12:06:32] DEBUG[18522]: chan_zap.c:1402 update_conf: Updated conferencing on 3, with 0 conference users [Mar 10 12:06:32] -- Registered channel 3, FXS Kewlstart signalling [Mar 10 12:06:32] DEBUG[18522]: chan_zap.c:1402 update_conf: Updated conferencing on 4, with 0 conference users [Mar 10 12:06:32] -- Registered channel 4, FXS Kewlstart signalling [Mar 10 12:06:32] DEBUG[18522]: chan_zap.c:1402 update_conf: Updated conferencing on 5, with 0 conference users [Mar 10 12:06:32] -- Registered channel 5, FXS Kewlstart signalling [Mar 10 12:06:32] DEBUG[18522]: chan_zap.c:1402 update_conf: Updated conferencing on 6, with 0 conference users [Mar 10 12:06:32] -- Registered channel 6, FXS Kewlstart signalling [Mar 10 12:06:32] DEBUG[18522]: chan_zap.c:1402 update_conf: Updated conferencing on 7, with 0 conference users [Mar 10 12:06:32] -- Registered channel 7, FXS Kewlstart signalling [Mar 10 12:06:32] DEBUG[18522]: chan_zap.c:1402 update_conf: Updated conferencing on 8, with 0 conference users [Mar 10 12:06:32] -- Registered channel 8, FXS Kewlstart signalling [Mar 10 12:06:32] -- Automatically generated pseudo channel [Mar 10 12:06:32] DEBUG[18522]: channel.c:524 ast_channel_register: Registered handler for 'Zap' (Zapata Telephony Driver w/PRI) [Mar 10 12:06:32] == Registered channel type 'Zap' (Zapata Telephony Driver w/PRI) [Mar 10 12:06:32] == Manager registered action ZapTransfer [Mar 10 12:06:32] == Manager registered action ZapHangup [Mar 10 12:06:32] == Manager registered action ZapDialOffhook [Mar 10 12:06:32] == Manager registered action ZapDNDon [Mar 10 12:06:32] == Manager registered action ZapDNDoff [Mar 10 12:06:32] == Manager registered action ZapShowChannels [Mar 10 12:06:32] == Manager registered action ZapRestart [Mar 10 12:06:32] chan_zap.so => (Zapata Telephony) [Mar 10 12:06:32] == Registered application 'ControlPlayback' [Mar 10 12:06:32] app_controlplayback.so => (Control Playback Application) [Mar 10 12:06:32] == Registered custom function URIDECODE [Mar 10 12:06:32] == Registered custom function URIENCODE [Mar 10 12:06:32] func_uri.so => (URI encode/decode dialplan functions) [Mar 10 12:06:32] == Registered custom function FIELDQTY [Mar 10 12:06:32] == Registered custom function FILTER [Mar 10 12:06:32] == Registered custom function REGEX [Mar 10 12:06:32] == Registered custom function ARRAY [Mar 10 12:06:32] == Registered custom function QUOTE [Mar 10 12:06:32] == Registered custom function LEN [Mar 10 12:06:32] == Registered custom function STRFTIME [Mar 10 12:06:32] == Registered custom function STRPTIME [Mar 10 12:06:32] == Registered custom function EVAL [Mar 10 12:06:32] == Registered custom function KEYPADHASH [Mar 10 12:06:32] == Registered custom function SPRINTF [Mar 10 12:06:32] func_strings.so => (String handling dialplan functions) [Mar 10 12:06:32] == Registered application 'ChanSpy' [Mar 10 12:06:32] == Registered application 'ExtenSpy' [Mar 10 12:06:32] app_chanspy.so => (Listen to the audio of an active channel) [Mar 10 12:06:32] == Registered application 'SetTransferCapability' [Mar 10 12:06:32] app_settransfercapability.so => (Set ISDN Transfer Capability) [Mar 10 12:06:32] == Registered application 'Pickup' [Mar 10 12:06:32] app_directed_pickup.so => (Directed Call Pickup Application) [Mar 10 12:06:32] == Registered application 'NBScat' [Mar 10 12:06:32] app_nbscat.so => (Silly NBS Stream Application) [Mar 10 12:06:32] == Registered custom function TIMEOUT [Mar 10 12:06:32] func_timeout.so => (Channel timeout dialplan functions) [Mar 10 12:06:32] == Registered custom function CALLERID [Mar 10 12:06:32] func_callerid.so => (Caller ID related dialplan function) [Mar 10 12:06:32] == Registered application 'VoiceMail' [Mar 10 12:06:32] == Registered application 'VoiceMailMain' [Mar 10 12:06:32] == Registered application 'MailboxExists' [Mar 10 12:06:32] == Registered application 'VMAuthenticate' [Mar 10 12:06:32] == Parsing '/etc/asterisk/voicemail.conf': [Mar 10 12:06:32] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/voicemail.conf [Mar 10 12:06:32] Found [Mar 10 12:06:32] DEBUG[18522]: app_voicemail.c:7368 load_config: VM Review Option disabled globally [Mar 10 12:06:32] DEBUG[18522]: app_voicemail.c:7376 load_config: VM Temperary Greeting Reminder Option disabled globally [Mar 10 12:06:32] DEBUG[18522]: app_voicemail.c:7386 load_config: VM Operator break disabled globally [Mar 10 12:06:32] DEBUG[18522]: app_voicemail.c:7393 load_config: VM CID Info before msg disabled globally [Mar 10 12:06:32] DEBUG[18522]: app_voicemail.c:7407 load_config: ENVELOPE before msg enabled globally [Mar 10 12:06:32] DEBUG[18522]: app_voicemail.c:7414 load_config: Duration info before msg enabled globally [Mar 10 12:06:32] DEBUG[18522]: app_voicemail.c:7430 load_config: We are not going to skip to the next msg after save/delete [Mar 10 12:06:32] app_voicemail.so => (Comedian Mail (Voicemail System)) [Mar 10 12:06:32] == Registered application 'ExternalIVR' [Mar 10 12:06:32] app_externalivr.so => (External IVR Interface Application) [Mar 10 12:06:32] == Registered custom function CHANNEL [Mar 10 12:06:32] func_channel.so => (Channel information dialplan function) [Mar 10 12:06:32] == Registered application 'MP3Player' [Mar 10 12:06:32] app_mp3.so => (Silly MP3 Application) [Mar 10 12:06:32] == Registered application 'ReadFile' [Mar 10 12:06:32] app_readfile.so => (Stores output of file into a variable) [Mar 10 12:06:32] == Registered application 'SendImage' [Mar 10 12:06:32] app_image.so => (Image Transmission Application) [Mar 10 12:06:32] == Registered application 'While' [Mar 10 12:06:32] == Registered application 'EndWhile' [Mar 10 12:06:32] == Registered application 'ExitWhile' [Mar 10 12:06:32] == Registered application 'ContinueWhile' [Mar 10 12:06:32] app_while.so => (While Loops and Conditional Execution) [Mar 10 12:06:32] == Registered application 'SayUnixTime' [Mar 10 12:06:32] == Registered application 'DateTime' [Mar 10 12:06:32] app_sayunixtime.so => (Say time) [Mar 10 12:06:32] == Registered custom function BASE64_ENCODE [Mar 10 12:06:32] == Registered custom function BASE64_DECODE [Mar 10 12:06:32] func_base64.so => (base64 encode/decode dialplan functions) [Mar 10 12:06:32] ERROR[18522]: app_amd.c:329 load_config: Configuration file amd.conf missing. [Mar 10 12:06:32] == Registered application 'AMD' [Mar 10 12:06:32] app_amd.so => (Answering Machine Detection Application) [Mar 10 12:06:32] == Registered application 'WaitForRing' [Mar 10 12:06:32] app_waitforring.so => (Waits until first ring after time) [Mar 10 12:06:32] == Registered custom function LANGUAGE [Mar 10 12:06:32] func_language.so => (Channel language dialplan function) [Mar 10 12:06:32] == Registered application 'Zapateller' [Mar 10 12:06:32] app_zapateller.so => (Block Telemarketers with Special Information Tone) [Mar 10 12:06:32] == Registered application 'NoCDR' [Mar 10 12:06:32] app_cdr.so => (Tell Asterisk to not maintain a CDR for the current call) [Mar 10 12:06:32] == Registered application 'Playback' [Mar 10 12:06:32] app_playback.so => (Sound File Playback Application) [Mar 10 12:06:32] == Manager registered action PlayDTMF [Mar 10 12:06:32] == Registered application 'SendDTMF' [Mar 10 12:06:32] app_senddtmf.so => (Send DTMF digits Application) [Mar 10 12:06:32] == Registered application 'ForkCDR' [Mar 10 12:06:32] app_forkcdr.so => (Fork The CDR into 2 separate entities) [Mar 10 12:06:32] == Registered application 'UserEvent' [Mar 10 12:06:32] app_userevent.so => (Custom User Event Application) [Mar 10 12:06:32] == Registered application 'TrySystem' [Mar 10 12:06:32] == Registered application 'System' [Mar 10 12:06:32] app_system.so => (Generic System() application) [Mar 10 12:06:32] == Registered translator 'alawtoulaw' from format alaw to ulaw, cost 1 [Mar 10 12:06:32] DEBUG[18522]: translate.c:425 rebuild_matrix: Resetting translation matrix [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to ulaw, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from g723 to unknown, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to ulaw, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from ulaw to g723, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from ulaw to unknown, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from ulaw to unknown, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from ulaw to unknown, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to gsm, via 3 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 4 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to gsm, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:32] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to gsm, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to gsm, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from g723 to gsm, via 3 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from g723 to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 6 cost path from ulaw to gsm, via 1 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from ulaw to gsm, via 3 [Mar 10 12:06:33] == Registered translator 'ulawtoalaw' from format ulaw to alaw, cost 1 [Mar 10 12:06:33] DEBUG[18522]: translate.c:425 rebuild_matrix: Resetting translation matrix [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to ulaw, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from g723 to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from gsm to unknown, via 3 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from gsm to g723, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from gsm to ulaw, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from gsm to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 16 cost path from gsm to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from gsm to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 2 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to ulaw, via 2 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 6 cost path from unknown to unknown, via 2 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 17 cost path from unknown to unknown, via 2 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 6 cost path from unknown to unknown, via 2 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to ulaw, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from ulaw to g723, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from ulaw to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from ulaw to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from ulaw to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to gsm, via 3 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to gsm, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to gsm, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to gsm, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from g723 to gsm, via 3 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from g723 to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from gsm to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 6 cost path from ulaw to gsm, via 1 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from ulaw to gsm, via 3 [Mar 10 12:06:33] codec_a_mu.so => (A-law and Mulaw direct Coder/Decoder) [Mar 10 12:06:33] == Registered file format h263, extension(s) h263 [Mar 10 12:06:33] format_h263.so => (Raw H.263 data) [Mar 10 12:06:33] == Registered file format vox, extension(s) vox [Mar 10 12:06:33] format_vox.so => (Dialogic VOX (ADPCM) File Format) [Mar 10 12:06:33] == Registered application 'BackgroundDetect' [Mar 10 12:06:33] app_talkdetect.so => (Playback with Talk Detection) [Mar 10 12:06:33] == Registered application 'SpeechCreate' [Mar 10 12:06:33] == Registered application 'SpeechLoadGrammar' [Mar 10 12:06:33] == Registered application 'SpeechUnloadGrammar' [Mar 10 12:06:33] == Registered application 'SpeechActivateGrammar' [Mar 10 12:06:33] == Registered application 'SpeechDeactivateGrammar' [Mar 10 12:06:33] == Registered application 'SpeechStart' [Mar 10 12:06:33] == Registered application 'SpeechBackground' [Mar 10 12:06:33] == Registered application 'SpeechDestroy' [Mar 10 12:06:33] == Registered application 'SpeechProcessingSound' [Mar 10 12:06:33] == Registered custom function SPEECH [Mar 10 12:06:33] == Registered custom function SPEECH_SCORE [Mar 10 12:06:33] == Registered custom function SPEECH_TEXT [Mar 10 12:06:33] == Registered custom function SPEECH_GRAMMAR [Mar 10 12:06:33] == Registered custom function SPEECH_ENGINE [Mar 10 12:06:33] app_speech_utils.so => (Dialplan Speech Applications) [Mar 10 12:06:33] == Parsing '/etc/asterisk/codecs.conf': [Mar 10 12:06:33] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/codecs.conf [Mar 10 12:06:33] Found [Mar 10 12:06:33] -- codec_adpcm: using generic PLC [Mar 10 12:06:33] WARNING[18522]: translate.c:675 __ast_register_translator: plc_samples 160 format 6 [Mar 10 12:06:33] == Registered translator 'adpcmtolin' from format adpcm to slin, cost 1 [Mar 10 12:06:33] DEBUG[18522]: translate.c:425 rebuild_matrix: Resetting translation matrix [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to ulaw, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from g723 to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from gsm to unknown, via 3 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from gsm to g723, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from gsm to ulaw, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from gsm to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 16 cost path from gsm to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from gsm to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 2 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to ulaw, via 2 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 6 cost path from unknown to unknown, via 2 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 17 cost path from unknown to unknown, via 2 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 6 cost path from unknown to unknown, via 2 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to ulaw, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from ulaw to g723, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from ulaw to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from ulaw to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from ulaw to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to ulaw, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to gsm, via 3 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to gsm, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to gsm, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to gsm, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from g723 to gsm, via 3 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from g723 to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from gsm to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 6 cost path from ulaw to gsm, via 1 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from ulaw to gsm, via 3 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 6 cost path from unknown to gsm, via 1 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to gsm, via 3 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 4 [Mar 10 12:06:33] == Registered translator 'lintoadpcm' from format slin to adpcm, cost 1 [Mar 10 12:06:33] DEBUG[18522]: translate.c:425 rebuild_matrix: Resetting translation matrix [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to ulaw, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from g723 to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from gsm to unknown, via 3 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from gsm to g723, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from gsm to ulaw, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from gsm to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from gsm to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 16 cost path from gsm to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from gsm to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to g723, via 2 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to ulaw, via 2 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 2 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 6 cost path from unknown to unknown, via 2 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 17 cost path from unknown to unknown, via 2 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 6 cost path from unknown to unknown, via 2 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to ulaw, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from ulaw to g723, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from ulaw to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from ulaw to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from ulaw to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from ulaw to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to ulaw, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to gsm, via 3 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to gsm, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to gsm, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to gsm, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from g723 to gsm, via 3 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from g723 to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from gsm to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 6 cost path from ulaw to gsm, via 1 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from ulaw to gsm, via 3 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 6 cost path from unknown to gsm, via 1 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to gsm, via 3 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 4 [Mar 10 12:06:33] codec_adpcm.so => (Adaptive Differential PCM Coder/Decoder) [Mar 10 12:06:33] == Parsing '/etc/asterisk/oss.conf': [Mar 10 12:06:33] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/oss.conf [Mar 10 12:06:33] Found [Mar 10 12:06:33] DEBUG[18522]: channel.c:524 ast_channel_register: Registered handler for 'Console' (OSS Console Channel Driver) [Mar 10 12:06:33] == Registered channel type 'Console' (OSS Console Channel Driver) [Mar 10 12:06:33] chan_oss.so => (OSS Console Channel Driver) [Mar 10 12:06:33] == Parsing '/etc/asterisk/codecs.conf': [Mar 10 12:06:33] DEBUG[18522]: config.c:851 config_text_file_load: Parsing /etc/asterisk/codecs.conf [Mar 10 12:06:33] Found [Mar 10 12:06:33] -- codec_ulaw: using generic PLC [Mar 10 12:06:33] WARNING[18522]: translate.c:675 __ast_register_translator: plc_samples 160 format 6 [Mar 10 12:06:33] == Registered translator 'ulawtolin' from format ulaw to slin, cost 1 [Mar 10 12:06:33] DEBUG[18522]: translate.c:425 rebuild_matrix: Resetting translation matrix [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to ulaw, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from g723 to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from gsm to g723, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from gsm to ulaw, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from gsm to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from gsm to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from gsm to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from gsm to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 2 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 2 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 2 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 2 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 2 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 2 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to ulaw, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from ulaw to g723, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from ulaw to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from ulaw to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from ulaw to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from ulaw to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to ulaw, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to gsm, via 3 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to gsm, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to gsm, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to gsm, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from g723 to gsm, via 3 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from g723 to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from gsm to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 6 cost path from ulaw to gsm, via 1 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from ulaw to gsm, via 3 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 6 cost path from unknown to gsm, via 1 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to gsm, via 3 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 4 [Mar 10 12:06:33] == Registered translator 'lintoulaw' from format slin to ulaw, cost 1 [Mar 10 12:06:33] DEBUG[18522]: translate.c:425 rebuild_matrix: Resetting translation matrix [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to gsm, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to ulaw, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from g723 to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from g723 to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from g723 to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from gsm to g723, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from gsm to ulaw, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from gsm to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from gsm to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from gsm to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from gsm to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 2 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 2 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 2 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 2 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 2 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 2 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to ulaw, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from ulaw to g723, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from ulaw to gsm, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from ulaw to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from ulaw to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from ulaw to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from ulaw to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to gsm, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to ulaw, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to gsm, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to gsm, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to ulaw, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to g723, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to gsm, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 5 cost path from unknown to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 16 cost path from unknown to unknown, via 4 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to g723, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to gsm, via 6 [Mar 10 12:06:33] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Mar 10 12:06:38] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 2 cost path from unknown to unknown, via 6 [Mar 10 12:06:38] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 4 cost path from unknown to unknown, via 6 [Mar 10 12:06:33] DEBUG[18552]: iax2-provision.c:253 iax_provision_version: Unable to create provisioning packet for 'si-000fd3000124' [Mar 10 12:06:38] DEBUG[18544]: chan_iax2.c:6538 socket_process: Packet arrived out of order (expecting 1, got 0) (frametype = 6, subclass = 13) [Mar 10 12:06:38] DEBUG[18544]: chan_iax2.c:6545 socket_process: Acking anyway [Mar 10 12:06:33] DEBUG[18555]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/551 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = Found Their Call ID: 55ae66476793@24.123.23.170 Their Tag 8008f64acdedd308808ed20d8e710732 Our tag: as28e6a2b1 [Mar 10 12:06:38] DEBUG[18546]: chan_iax2.c:6538 socket_process: Packet arrived out of order (expecting 1, got 0) (frametype = 6, subclass = 13) [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:38] DEBUG[18546]: chan_iax2.c:6545 socket_process: Acking anyway [Mar 10 12:06:38] DEBUG[18543]: chan_iax2.c:6538 socket_process: Packet arrived out of order (expecting 1, got 0) (frametype = 6, subclass = 13) [Mar 10 12:06:38] DEBUG[18543]: chan_iax2.c:6545 socket_process: Acking anyway [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = Found Their Call ID: 55ae66476793@24.123.23.170 Their Tag 8008f64acdedd308808ed20d8e710732 Our tag: as28e6a2b1 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 39008, ours 39008) [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = Found Their Call ID: 55ae66476793@24.123.23.170 Their Tag 8008f64acdedd308808ed20d8e710732 Our tag: as28e6a2b1 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 39008, ours 39008) [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66476793@24.123.23.170 Their Tag 8008f64acdedd308808ed20d8e710732 Our tag: as28e6a2b1 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae664696c0@24.123.23.170 Their Tag 93cc73e47e62b2ad602ae1a53505884b Our tag: as7d71073d [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 1319586744@192.168.1.97 Their Tag 4210231806 Our tag: as609501e8 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66469f7f@24.123.23.170 Their Tag ad4bac0c41ca9130ca8935d73f9d9cfb Our tag: as745f2deb [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = Found Their Call ID: 2c01f4fc3cc0317e799f35d0601622e1@24.123.23.170 Their Tag Our tag: as005b8eb2 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '2c01f4fc3cc0317e799f35d0601622e1@24.123.23.170' of Request 103: Match Found [Mar 10 12:06:38] WARNING[18555]: chan_sip.c:12015 handle_response_register: Got 200 OK on REGISTER that isn't a register [Mar 10 12:06:38] Really destroying SIP dialog '2c01f4fc3cc0317e799f35d0601622e1@24.123.23.170' Method: REGISTER [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66476793@24.123.23.170 Their Tag 8008f64acdedd308808ed20d8e710732 Our tag: as28e6a2b1 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae664696c0@24.123.23.170 Their Tag 93cc73e47e62b2ad602ae1a53505884b Our tag: as7d71073d [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 1319586744@192.168.1.97 Their Tag 4210231806 Our tag: as609501e8 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66469f7f@24.123.23.170 Their Tag ad4bac0c41ca9130ca8935d73f9d9cfb Our tag: as745f2deb [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:14774 sipsock_read: Invalid SIP message - rejected , no callid, len 351 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66476793@24.123.23.170 Their Tag 8008f64acdedd308808ed20d8e710732 Our tag: as28e6a2b1 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae664696c0@24.123.23.170 Their Tag 93cc73e47e62b2ad602ae1a53505884b Our tag: as7d71073d [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 1319586744@192.168.1.97 Their Tag 4210231806 Our tag: as609501e8 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66469f7f@24.123.23.170 Their Tag ad4bac0c41ca9130ca8935d73f9d9cfb Our tag: as745f2deb [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4308 sip_alloc: Allocating new SIP dialog for 55ae6647678c@24.123.23.170 - REGISTER (No RTP) [Mar 10 12:06:38] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 15 cost path from unknown to unknown, via 6 [Mar 10 12:06:38] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 551 [Mar 10 12:06:38] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 551 [Mar 10 12:06:38] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/551 - state 1 (Not in use) [Mar 10 12:06:38] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from g723 to unknown, via 4 [Mar 10 12:06:38] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from gsm to unknown, via 4 [Mar 10 12:06:38] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 4 [Mar 10 12:06:38] DEBUG[18522]: translate.c:480 rebuild_matrix: Discovered 3 cost path from unknown to unknown, via 4 [Mar 10 12:06:38] DEBUG[18564]: app_queue.c:546 changethread: Device 'SIP/551' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Mar 10 12:06:38] codec_ulaw.so => (mu-Law Coder/Decoder) [Mar 10 12:06:38] == Registered custom function BLACKLIST [Mar 10 12:06:38] == Registered application 'LookupBlacklist' [Mar 10 12:06:38] app_lookupblacklist.so => (Look up Caller*ID name/number from blacklist database) [Mar 10 12:06:38] == Registered application 'GetCPEID' [Mar 10 12:06:38] app_getcpeid.so => (Get ADSI CPE ID) [Mar 10 12:06:38] pbx_realtime.so => (Realtime Switch) [Mar 10 12:06:38] == Registered file format wav, extension(s) wav [Mar 10 12:06:38] format_wav.so => (Microsoft WAV format (8000Hz Signed Linear)) [Mar 10 12:06:38] == Registered application 'ChannelRedirect' [Mar 10 12:06:38] app_channelredirect.so => (Channel Redirect) [Mar 10 12:06:38] == Registered application 'SMS' [Mar 10 12:06:38] app_sms.so => (SMS/PSTN handler) [Mar 10 12:06:38] == Registered custom function ENV [Mar 10 12:06:38] == Registered custom function STAT [Mar 10 12:06:38] func_env.so => (Environment/filesystem dialplan functions) [Mar 10 12:06:38] res_clioriginate.so => (Call origination from the CLI) [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = Found Their Call ID: 55ae6647678c@24.123.23.170 Their Tag 3ecae86c4b21a3e38a3d79522e6f3263 Our tag: as388eb9f0 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 61342, ours 61342) [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae6647678c@24.123.23.170 Their Tag 3ecae86c4b21a3e38a3d79522e6f3263 Our tag: as388eb9f0 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = Found Their Call ID: 55ae66476793@24.123.23.170 Their Tag 8008f64acdedd308808ed20d8e710732 Our tag: as28e6a2b1 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 39008, ours 39008) [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = Found Their Call ID: 55ae6647678c@24.123.23.170 Their Tag 3ecae86c4b21a3e38a3d79522e6f3263 Our tag: as388eb9f0 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 61342, ours 61342) [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae6647678c@24.123.23.170 Their Tag 3ecae86c4b21a3e38a3d79522e6f3263 Our tag: as388eb9f0 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66476793@24.123.23.170 Their Tag 8008f64acdedd308808ed20d8e710732 Our tag: as28e6a2b1 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae664696c0@24.123.23.170 Their Tag 93cc73e47e62b2ad602ae1a53505884b Our tag: as7d71073d [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 1319586744@192.168.1.97 Their Tag 4210231806 Our tag: as609501e8 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66469f7f@24.123.23.170 Their Tag ad4bac0c41ca9130ca8935d73f9d9cfb Our tag: as745f2deb [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4308 sip_alloc: Allocating new SIP dialog for 55ae66469f11@24.123.23.170 - REGISTER (No RTP) [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = Found Their Call ID: 55ae66469f11@24.123.23.170 Their Tag 3e06c3fb94b5c10174c169327f26f5af Our tag: as2c9c5044 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 49841, ours 49841) [Mar 10 12:06:38] == Registered application 'Authenticate' [Mar 10 12:06:38] app_authenticate.so => (Authentication Application) [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66469f11@24.123.23.170 Their Tag 3e06c3fb94b5c10174c169327f26f5af Our tag: as2c9c5044 [Mar 10 12:06:38] Asterisk Ready. [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae6647678c@24.123.23.170 Their Tag 3ecae86c4b21a3e38a3d79522e6f3263 Our tag: as388eb9f0 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66476793@24.123.23.170 Their Tag 8008f64acdedd308808ed20d8e710732 Our tag: as28e6a2b1 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae664696c0@24.123.23.170 Their Tag 93cc73e47e62b2ad602ae1a53505884b Our tag: as7d71073d [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 1319586744@192.168.1.97 Their Tag 4210231806 Our tag: as609501e8 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66469f7f@24.123.23.170 Their Tag ad4bac0c41ca9130ca8935d73f9d9cfb Our tag: as745f2deb [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4308 sip_alloc: Allocating new SIP dialog for 55ae6647666e@24.123.23.170 - REGISTER (No RTP) [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER ]1;Asterisk]2;Asterisk Console on 'unifiedpaging.messagenetsystems.com' (pid 18522)*CLI> [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae6647666e@24.123.23.170 Their Tag a0d0dbc182dc6557fdc044be4ef5ea6a Our tag: as25c37ecb [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66469f11@24.123.23.170 Their Tag 3e06c3fb94b5c10174c169327f26f5af Our tag: as2c9c5044 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = Found Their Call ID: 55ae6647678c@24.123.23.170 Their Tag 3ecae86c4b21a3e38a3d79522e6f3263 Our tag: as388eb9f0 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 61342, ours 61342) [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = Found Their Call ID: 55ae6647666e@24.123.23.170 Their Tag a0d0dbc182dc6557fdc044be4ef5ea6a Our tag: as25c37ecb [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 24033, ours 24033) [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae6647666e@24.123.23.170 Their Tag a0d0dbc182dc6557fdc044be4ef5ea6a Our tag: as25c37ecb [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = Found Their Call ID: 55ae66469f11@24.123.23.170 Their Tag 3e06c3fb94b5c10174c169327f26f5af Our tag: as2c9c5044 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 49841, ours 49841) [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae6647666e@24.123.23.170 Their Tag a0d0dbc182dc6557fdc044be4ef5ea6a Our tag: as25c37ecb [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66469f11@24.123.23.170 Their Tag 3e06c3fb94b5c10174c169327f26f5af Our tag: as2c9c5044 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae6647678c@24.123.23.170 Their Tag 3ecae86c4b21a3e38a3d79522e6f3263 Our tag: as388eb9f0 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = Found Their Call ID: 55ae66476793@24.123.23.170 Their Tag 8008f64acdedd308808ed20d8e710732 Our tag: as28e6a2b1 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = Found Their Call ID: 55ae6647666e@24.123.23.170 Their Tag a0d0dbc182dc6557fdc044be4ef5ea6a Our tag: as25c37ecb [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae6647666e@24.123.23.170 Their Tag a0d0dbc182dc6557fdc044be4ef5ea6a Our tag: as25c37ecb [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = Found Their Call ID: 55ae66469f11@24.123.23.170 Their Tag 3e06c3fb94b5c10174c169327f26f5af Our tag: as2c9c5044 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae6647666e@24.123.23.170 Their Tag a0d0dbc182dc6557fdc044be4ef5ea6a Our tag: as25c37ecb [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66469f11@24.123.23.170 Their Tag 3e06c3fb94b5c10174c169327f26f5af Our tag: as2c9c5044 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = Found Their Call ID: 55ae6647678c@24.123.23.170 Their Tag 3ecae86c4b21a3e38a3d79522e6f3263 Our tag: as388eb9f0 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae6647666e@24.123.23.170 Their Tag a0d0dbc182dc6557fdc044be4ef5ea6a Our tag: as25c37ecb [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66469f11@24.123.23.170 Their Tag 3e06c3fb94b5c10174c169327f26f5af Our tag: as2c9c5044 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae6647678c@24.123.23.170 Their Tag 3ecae86c4b21a3e38a3d79522e6f3263 Our tag: as388eb9f0 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66476793@24.123.23.170 Their Tag 8008f64acdedd308808ed20d8e710732 Our tag: as28e6a2b1 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae664696c0@24.123.23.170 Their Tag 93cc73e47e62b2ad602ae1a53505884b Our tag: as7d71073d [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 1319586744@192.168.1.97 Their Tag 4210231806 Our tag: as609501e8 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66469f7f@24.123.23.170 Their Tag ad4bac0c41ca9130ca8935d73f9d9cfb Our tag: as745f2deb [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4308 sip_alloc: Allocating new SIP dialog for 000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165 - REGISTER (No RTP) [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = Found Their Call ID: 000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165 Their Tag Our tag: as2b55c95b [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:38] -- Saved useragent "CSCO/7" for peer 514 [Mar 10 12:06:38] DEBUG[18555]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/514 [Mar 10 12:06:38] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 514 [Mar 10 12:06:38] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 514 [Mar 10 12:06:38] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/514 - state 1 (Not in use) [Mar 10 12:06:38] DEBUG[18566]: 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. [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165 Their Tag Our tag: as2b55c95b [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = Found Their Call ID: 55ae6647666e@24.123.23.170 Their Tag a0d0dbc182dc6557fdc044be4ef5ea6a Our tag: as25c37ecb [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:38] -- Saved useragent "Uniden SIP Phone p2 Ver BS4.77" for peer 526 [Mar 10 12:06:38] DEBUG[18555]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/526 [Mar 10 12:06:38] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 526 [Mar 10 12:06:38] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 526 [Mar 10 12:06:38] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/526 - state 1 (Not in use) [Mar 10 12:06:38] DEBUG[18567]: 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. [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165 Their Tag Our tag: as2b55c95b [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae6647666e@24.123.23.170 Their Tag a0d0dbc182dc6557fdc044be4ef5ea6a Our tag: as25c37ecb [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = Found Their Call ID: 55ae66469f11@24.123.23.170 Their Tag 3e06c3fb94b5c10174c169327f26f5af Our tag: as2c9c5044 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:38] -- Saved useragent "Uniden SIP Phone p2 Ver BS4.77" for peer 522 [Mar 10 12:06:38] DEBUG[18555]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/522 [Mar 10 12:06:38] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 522 [Mar 10 12:06:38] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 522 [Mar 10 12:06:38] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/522 - state 1 (Not in use) [Mar 10 12:06:38] DEBUG[18568]: 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. [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165 Their Tag Our tag: as2b55c95b [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae6647666e@24.123.23.170 Their Tag a0d0dbc182dc6557fdc044be4ef5ea6a Our tag: as25c37ecb [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66469f11@24.123.23.170 Their Tag 3e06c3fb94b5c10174c169327f26f5af Our tag: as2c9c5044 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae6647678c@24.123.23.170 Their Tag 3ecae86c4b21a3e38a3d79522e6f3263 Our tag: as388eb9f0 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = Found Their Call ID: 55ae66476793@24.123.23.170 Their Tag 8008f64acdedd308808ed20d8e710732 Our tag: as28e6a2b1 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:38] -- Saved useragent "Uniden SIP Phone p2 Ver BS4.77" for peer 525 [Mar 10 12:06:38] DEBUG[18555]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/525 [Mar 10 12:06:38] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 525 [Mar 10 12:06:38] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 525 [Mar 10 12:06:38] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/525 - state 1 (Not in use) [Mar 10 12:06:38] DEBUG[18569]: 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. [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165 Their Tag Our tag: as2b55c95b [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae6647666e@24.123.23.170 Their Tag a0d0dbc182dc6557fdc044be4ef5ea6a Our tag: as25c37ecb [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66469f11@24.123.23.170 Their Tag 3e06c3fb94b5c10174c169327f26f5af Our tag: as2c9c5044 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:4359 find_call: = Found Their Call ID: 55ae6647678c@24.123.23.170 Their Tag 3ecae86c4b21a3e38a3d79522e6f3263 Our tag: as388eb9f0 [Mar 10 12:06:38] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:38] -- Saved useragent "Uniden SIP Phone p2 Ver BS4.77" for peer 524 [Mar 10 12:06:38] DEBUG[18555]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/524 [Mar 10 12:06:38] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 524 [Mar 10 12:06:38] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 524 [Mar 10 12:06:38] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/524 - state 1 (Not in use) [Mar 10 12:06:38] DEBUG[18570]: app_queue.c:546 changethread: Device 'SIP/524' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Mar 10 12:06:39] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165 Their Tag Our tag: as2b55c95b [Mar 10 12:06:39] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae6647666e@24.123.23.170 Their Tag a0d0dbc182dc6557fdc044be4ef5ea6a Our tag: as25c37ecb [Mar 10 12:06:39] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66469f11@24.123.23.170 Their Tag 3e06c3fb94b5c10174c169327f26f5af Our tag: as2c9c5044 [Mar 10 12:06:39] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae6647678c@24.123.23.170 Their Tag 3ecae86c4b21a3e38a3d79522e6f3263 Our tag: as388eb9f0 [Mar 10 12:06:39] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66476793@24.123.23.170 Their Tag 8008f64acdedd308808ed20d8e710732 Our tag: as28e6a2b1 [Mar 10 12:06:39] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae664696c0@24.123.23.170 Their Tag 93cc73e47e62b2ad602ae1a53505884b Our tag: as7d71073d [Mar 10 12:06:39] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 1319586744@192.168.1.97 Their Tag 4210231806 Our tag: as609501e8 [Mar 10 12:06:39] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66469f7f@24.123.23.170 Their Tag ad4bac0c41ca9130ca8935d73f9d9cfb Our tag: as745f2deb [Mar 10 12:06:39] DEBUG[18555]: chan_sip.c:4308 sip_alloc: Allocating new SIP dialog for 000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165 - REGISTER (No RTP) [Mar 10 12:06:39] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:39] DEBUG[18555]: chan_sip.c:4359 find_call: = Found Their Call ID: 000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165 Their Tag Our tag: as040f99ce [Mar 10 12:06:39] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:39] -- Saved useragent "CSCO/7" for peer 513 [Mar 10 12:06:39] DEBUG[18555]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/513 [Mar 10 12:06:39] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 513 [Mar 10 12:06:39] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 513 [Mar 10 12:06:39] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/513 - state 1 (Not in use) [Mar 10 12:06:39] DEBUG[18571]: 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. [Mar 10 12:06:40] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165 Their Tag Our tag: as040f99ce [Mar 10 12:06:40] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165 Their Tag Our tag: as2b55c95b [Mar 10 12:06:40] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae6647666e@24.123.23.170 Their Tag a0d0dbc182dc6557fdc044be4ef5ea6a Our tag: as25c37ecb [Mar 10 12:06:40] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66469f11@24.123.23.170 Their Tag 3e06c3fb94b5c10174c169327f26f5af Our tag: as2c9c5044 [Mar 10 12:06:40] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae6647678c@24.123.23.170 Their Tag 3ecae86c4b21a3e38a3d79522e6f3263 Our tag: as388eb9f0 [Mar 10 12:06:40] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66476793@24.123.23.170 Their Tag 8008f64acdedd308808ed20d8e710732 Our tag: as28e6a2b1 [Mar 10 12:06:40] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae664696c0@24.123.23.170 Their Tag 93cc73e47e62b2ad602ae1a53505884b Our tag: as7d71073d [Mar 10 12:06:40] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 1319586744@192.168.1.97 Their Tag 4210231806 Our tag: as609501e8 [Mar 10 12:06:40] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66469f7f@24.123.23.170 Their Tag ad4bac0c41ca9130ca8935d73f9d9cfb Our tag: as745f2deb [Mar 10 12:06:40] DEBUG[18555]: chan_sip.c:4308 sip_alloc: Allocating new SIP dialog for 55ae664767e0@24.123.23.170 - REGISTER (No RTP) [Mar 10 12:06:40] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:40] DEBUG[18555]: chan_sip.c:4359 find_call: = Found Their Call ID: 55ae664767e0@24.123.23.170 Their Tag 611c0e1eb2b40cda141656493953c3aa Our tag: as1ba9a031 [Mar 10 12:06:40] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:40] -- Saved useragent "Uniden SIP Phone p2 Ver BS4.77" for peer 523 [Mar 10 12:06:40] DEBUG[18555]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/523 [Mar 10 12:06:40] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 523 [Mar 10 12:06:40] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 523 [Mar 10 12:06:40] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/523 - state 1 (Not in use) [Mar 10 12:06:40] DEBUG[18572]: 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. [Mar 10 12:06:42] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae664767e0@24.123.23.170 Their Tag 611c0e1eb2b40cda141656493953c3aa Our tag: as1ba9a031 [Mar 10 12:06:42] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165 Their Tag Our tag: as040f99ce [Mar 10 12:06:42] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165 Their Tag Our tag: as2b55c95b [Mar 10 12:06:42] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae6647666e@24.123.23.170 Their Tag a0d0dbc182dc6557fdc044be4ef5ea6a Our tag: as25c37ecb [Mar 10 12:06:42] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66469f11@24.123.23.170 Their Tag 3e06c3fb94b5c10174c169327f26f5af Our tag: as2c9c5044 [Mar 10 12:06:42] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae6647678c@24.123.23.170 Their Tag 3ecae86c4b21a3e38a3d79522e6f3263 Our tag: as388eb9f0 [Mar 10 12:06:42] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66476793@24.123.23.170 Their Tag 8008f64acdedd308808ed20d8e710732 Our tag: as28e6a2b1 [Mar 10 12:06:42] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae664696c0@24.123.23.170 Their Tag 93cc73e47e62b2ad602ae1a53505884b Our tag: as7d71073d [Mar 10 12:06:42] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 1319586744@192.168.1.97 Their Tag 4210231806 Our tag: as609501e8 [Mar 10 12:06:42] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66469f7f@24.123.23.170 Their Tag ad4bac0c41ca9130ca8935d73f9d9cfb Our tag: as745f2deb [Mar 10 12:06:42] DEBUG[18555]: chan_sip.c:4308 sip_alloc: Allocating new SIP dialog for 000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165 - REGISTER (No RTP) [Mar 10 12:06:42] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:42] DEBUG[18555]: chan_sip.c:4359 find_call: = Found Their Call ID: 000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165 Their Tag Our tag: as689d1970 [Mar 10 12:06:42] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:42] -- Saved useragent "CSCO/7" for peer 512 [Mar 10 12:06:42] DEBUG[18555]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/512 [Mar 10 12:06:42] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 512 [Mar 10 12:06:42] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 512 [Mar 10 12:06:42] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/512 - state 1 (Not in use) [Mar 10 12:06:42] DEBUG[18573]: 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. [Mar 10 12:06:42] DEBUG[18555]: chan_sip.c:4308 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Mar 10 12:06:43] DEBUG[18555]: chan_sip.c:4359 find_call: = Found Their Call ID: 1b0e7c567baaa224791ba96172aa9c1d@24.123.23.170 Their Tag Our tag: as7ca28428 [Mar 10 12:06:43] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '1b0e7c567baaa224791ba96172aa9c1d@24.123.23.170' of Request 102: Match Not Found [Mar 10 12:06:43] NOTICE[18555]: chan_sip.c:12120 handle_response_peerpoke: Peer 'Broadvoice' is now Reachable. (30ms / 2000ms) [Mar 10 12:06:43] DEBUG[18555]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/Broadvoice [Mar 10 12:06:43] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - Broadvoice [Mar 10 12:06:43] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer Broadvoice [Mar 10 12:06:43] Really destroying SIP dialog '1b0e7c567baaa224791ba96172aa9c1d@24.123.23.170' Method: OPTIONS [Mar 10 12:06:43] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/Broadvoice - state 1 (Not in use) [Mar 10 12:06:43] DEBUG[18577]: 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. [Mar 10 12:06:44] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165 Their Tag Our tag: as689d1970 [Mar 10 12:06:44] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae664767e0@24.123.23.170 Their Tag 611c0e1eb2b40cda141656493953c3aa Our tag: as1ba9a031 [Mar 10 12:06:44] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165 Their Tag Our tag: as040f99ce [Mar 10 12:06:44] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165 Their Tag Our tag: as2b55c95b [Mar 10 12:06:44] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae6647666e@24.123.23.170 Their Tag a0d0dbc182dc6557fdc044be4ef5ea6a Our tag: as25c37ecb [Mar 10 12:06:44] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66469f11@24.123.23.170 Their Tag 3e06c3fb94b5c10174c169327f26f5af Our tag: as2c9c5044 [Mar 10 12:06:44] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae6647678c@24.123.23.170 Their Tag 3ecae86c4b21a3e38a3d79522e6f3263 Our tag: as388eb9f0 [Mar 10 12:06:44] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66476793@24.123.23.170 Their Tag 8008f64acdedd308808ed20d8e710732 Our tag: as28e6a2b1 [Mar 10 12:06:44] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae664696c0@24.123.23.170 Their Tag 93cc73e47e62b2ad602ae1a53505884b Our tag: as7d71073d [Mar 10 12:06:44] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 1319586744@192.168.1.97 Their Tag 4210231806 Our tag: as609501e8 [Mar 10 12:06:44] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66469f7f@24.123.23.170 Their Tag ad4bac0c41ca9130ca8935d73f9d9cfb Our tag: as745f2deb [Mar 10 12:06:44] DEBUG[18555]: chan_sip.c:4308 sip_alloc: Allocating new SIP dialog for 000ff78d-ebb20007-4ad25281-15520cf9@192.168.1.165 - REGISTER (No RTP) [Mar 10 12:06:44] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:44] DEBUG[18555]: chan_sip.c:4359 find_call: = Found Their Call ID: 000ff78d-ebb20007-4ad25281-15520cf9@192.168.1.165 Their Tag Our tag: as4d4f8a7f [Mar 10 12:06:44] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:44] -- Saved useragent "CSCO/7" for peer 515 [Mar 10 12:06:44] DEBUG[18555]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/515 [Mar 10 12:06:44] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 515 [Mar 10 12:06:44] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 515 [Mar 10 12:06:44] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/515 - state 1 (Not in use) [Mar 10 12:06:44] DEBUG[18578]: 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. *CLI> [Mar 10 12:06:48] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 000ff78d-ebb20007-4ad25281-15520cf9@192.168.1.165 Their Tag Our tag: as4d4f8a7f [Mar 10 12:06:48] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165 Their Tag Our tag: as689d1970 [Mar 10 12:06:48] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae664767e0@24.123.23.170 Their Tag 611c0e1eb2b40cda141656493953c3aa Our tag: as1ba9a031 [Mar 10 12:06:48] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165 Their Tag Our tag: as040f99ce [Mar 10 12:06:48] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165 Their Tag Our tag: as2b55c95b [Mar 10 12:06:48] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae6647666e@24.123.23.170 Their Tag a0d0dbc182dc6557fdc044be4ef5ea6a Our tag: as25c37ecb [Mar 10 12:06:48] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66469f11@24.123.23.170 Their Tag 3e06c3fb94b5c10174c169327f26f5af Our tag: as2c9c5044 [Mar 10 12:06:48] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae6647678c@24.123.23.170 Their Tag 3ecae86c4b21a3e38a3d79522e6f3263 Our tag: as388eb9f0 [Mar 10 12:06:48] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66476793@24.123.23.170 Their Tag 8008f64acdedd308808ed20d8e710732 Our tag: as28e6a2b1 [Mar 10 12:06:48] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae664696c0@24.123.23.170 Their Tag 93cc73e47e62b2ad602ae1a53505884b Our tag: as7d71073d [Mar 10 12:06:48] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 1319586744@192.168.1.97 Their Tag 4210231806 Our tag: as609501e8 [Mar 10 12:06:48] DEBUG[18555]: chan_sip.c:4359 find_call: = Found Their Call ID: 55ae66469f7f@24.123.23.170 Their Tag ad4bac0c41ca9130ca8935d73f9d9cfb Our tag: as745f2deb [Mar 10 12:06:48] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:48] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 000ff78d-ebb20007-4ad25281-15520cf9@192.168.1.165 Their Tag Our tag: as4d4f8a7f [Mar 10 12:06:48] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165 Their Tag Our tag: as689d1970 [Mar 10 12:06:48] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae664767e0@24.123.23.170 Their Tag 611c0e1eb2b40cda141656493953c3aa Our tag: as1ba9a031 [Mar 10 12:06:48] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165 Their Tag Our tag: as040f99ce [Mar 10 12:06:48] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165 Their Tag Our tag: as2b55c95b [Mar 10 12:06:48] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae6647666e@24.123.23.170 Their Tag a0d0dbc182dc6557fdc044be4ef5ea6a Our tag: as25c37ecb [Mar 10 12:06:48] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66469f11@24.123.23.170 Their Tag 3e06c3fb94b5c10174c169327f26f5af Our tag: as2c9c5044 [Mar 10 12:06:48] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae6647678c@24.123.23.170 Their Tag 3ecae86c4b21a3e38a3d79522e6f3263 Our tag: as388eb9f0 [Mar 10 12:06:48] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66476793@24.123.23.170 Their Tag 8008f64acdedd308808ed20d8e710732 Our tag: as28e6a2b1 [Mar 10 12:06:48] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae664696c0@24.123.23.170 Their Tag 93cc73e47e62b2ad602ae1a53505884b Our tag: as7d71073d [Mar 10 12:06:48] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 1319586744@192.168.1.97 Their Tag 4210231806 Our tag: as609501e8 [Mar 10 12:06:48] DEBUG[18555]: chan_sip.c:4359 find_call: = Found Their Call ID: 55ae66469f7f@24.123.23.170 Their Tag ad4bac0c41ca9130ca8935d73f9d9cfb Our tag: as745f2deb [Mar 10 12:06:48] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:48] -- Saved useragent "Uniden SIP Phone p2 Ver BS4.77" for peer 528 [Mar 10 12:06:48] DEBUG[18555]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/528 [Mar 10 12:06:48] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 000ff78d-ebb20007-4ad25281-15520cf9@192.168.1.165 Their Tag Our tag: as4d4f8a7f [Mar 10 12:06:48] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165 Their Tag Our tag: as689d1970 [Mar 10 12:06:48] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae664767e0@24.123.23.170 Their Tag 611c0e1eb2b40cda141656493953c3aa Our tag: as1ba9a031 [Mar 10 12:06:48] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165 Their Tag Our tag: as040f99ce [Mar 10 12:06:48] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165 Their Tag Our tag: as2b55c95b [Mar 10 12:06:48] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae6647666e@24.123.23.170 Their Tag a0d0dbc182dc6557fdc044be4ef5ea6a Our tag: as25c37ecb [Mar 10 12:06:48] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66469f11@24.123.23.170 Their Tag 3e06c3fb94b5c10174c169327f26f5af Our tag: as2c9c5044 [Mar 10 12:06:48] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae6647678c@24.123.23.170 Their Tag 3ecae86c4b21a3e38a3d79522e6f3263 Our tag: as388eb9f0 [Mar 10 12:06:48] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66476793@24.123.23.170 Their Tag 8008f64acdedd308808ed20d8e710732 Our tag: as28e6a2b1 [Mar 10 12:06:48] DEBUG[18555]: chan_sip.c:4359 find_call: = Found Their Call ID: 55ae664696c0@24.123.23.170 Their Tag 93cc73e47e62b2ad602ae1a53505884b Our tag: as7d71073d [Mar 10 12:06:48] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:48] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 528 [Mar 10 12:06:48] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 528 [Mar 10 12:06:48] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/528 - state 1 (Not in use) [Mar 10 12:06:48] DEBUG[18579]: 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. [Mar 10 12:06:49] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 000ff78d-ebb20007-4ad25281-15520cf9@192.168.1.165 Their Tag Our tag: as4d4f8a7f [Mar 10 12:06:49] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165 Their Tag Our tag: as689d1970 [Mar 10 12:06:49] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae664767e0@24.123.23.170 Their Tag 611c0e1eb2b40cda141656493953c3aa Our tag: as1ba9a031 [Mar 10 12:06:49] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165 Their Tag Our tag: as040f99ce [Mar 10 12:06:49] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165 Their Tag Our tag: as2b55c95b [Mar 10 12:06:49] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae6647666e@24.123.23.170 Their Tag a0d0dbc182dc6557fdc044be4ef5ea6a Our tag: as25c37ecb [Mar 10 12:06:49] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66469f11@24.123.23.170 Their Tag 3e06c3fb94b5c10174c169327f26f5af Our tag: as2c9c5044 [Mar 10 12:06:49] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae6647678c@24.123.23.170 Their Tag 3ecae86c4b21a3e38a3d79522e6f3263 Our tag: as388eb9f0 [Mar 10 12:06:49] DEBUG[18555]: chan_sip.c:4359 find_call: = No match Their Call ID: 55ae66476793@24.123.23.170 Their Tag 8008f64acdedd308808ed20d8e710732 Our tag: as28e6a2b1 [Mar 10 12:06:49] DEBUG[18555]: chan_sip.c:4359 find_call: = Found Their Call ID: 55ae664696c0@24.123.23.170 Their Tag 93cc73e47e62b2ad602ae1a53505884b Our tag: as7d71073d [Mar 10 12:06:49] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:06:49] -- Saved useragent "Uniden SIP Phone p2 Ver BS4.77" for peer 529 [Mar 10 12:06:49] DEBUG[18555]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/529 [Mar 10 12:06:49] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 529 [Mar 10 12:06:49] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 529 [Mar 10 12:06:49] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/529 - state 1 (Not in use) [Mar 10 12:06:49] DEBUG[18580]: 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. set debug 4 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[Mar 10 12:06:56] DEBUG[18555]: chan_sip.c:4308 sip_alloc: Allocating new SIP dialog for 2c01f4fc3cc0317e799f35d0601622e1@24.123.23.170 - REGISTER (No RTP) [Mar 10 12:06:56] DEBUG[18555]: chan_sip.c:7287 transmit_register: Scheduled a registration timeout for sip.broadvoice.com id #154 [Mar 10 12:06:56] REGISTER attempt 1 to 3173241052@sip.broadvoice.com [Mar 10 12:06:57] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '2c01f4fc3cc0317e799f35d0601622e1@24.123.23.170' of Request 104: Match Not Found [Mar 10 12:06:57] DEBUG[18555]: chan_sip.c:12025 handle_response_register: Registration successful [Mar 10 12:06:57] DEBUG[18555]: chan_sip.c:12028 handle_response_register: Cancelling timeout 154 4 Core debug is at least 4 *CLI> core [Mar 10 12:07:00] DEBUG[18547]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel IAX2/614 [Mar 10 12:07:00] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for IAX2 - 614 [Mar 10 12:07:00] DEBUG[18527]: chan_iax2.c:9636 iax2_devicestate: Checking device state for device 614 [Mar 10 12:07:00] DEBUG[18527]: chan_iax2.c:9644 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 614? addr=-1493063488, defaddr=0 maxms=0, lastms=0 [Mar 10 12:07:00] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for IAX2/614 - state 1 (Not in use) [Mar 10 12:07:00] DEBUG[18581]: 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. [Mar 10 12:07:00] DEBUG[18545]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel IAX2/651 [Mar 10 12:07:00] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for IAX2 - 651 [Mar 10 12:07:00] DEBUG[18527]: chan_iax2.c:9636 iax2_devicestate: Checking device state for device 651 [Mar 10 12:07:00] DEBUG[18527]: chan_iax2.c:9644 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 651? addr=-1459509056, defaddr=0 maxms=0, lastms=0 [Mar 10 12:07:00] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for IAX2/651 - state 1 (Not in use) [Mar 10 12:07:00] DEBUG[18582]: 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. [Mar 10 12:07:03] DEBUG[18583]: manager.c:1973 process_message: Manager received command 'Login' [Mar 10 12:07:03] == Parsing '/etc/asterisk/manager.conf': [Mar 10 12:07:03] DEBUG[18583]: config.c:851 config_text_file_load: Parsing /etc/asterisk/manager.conf [Mar 10 12:07:03] Found [Mar 10 12:07:03] DEBUG[18583]: acl.c:200 ast_append_ha: 127.0.0.1/255.255.255.0/255.255.255.0 appended to acl for peer [Mar 10 12:07:03] DEBUG[18583]: acl.c:215 ast_apply_ha: ##### Testing 127.0.0.1 with 127.0.0.0 [Mar 10 12:07:03] == Manager 'MessageNet' logged on from 127.0.0.1 [Mar 10 12:07:03] DEBUG[18583]: manager.c:1973 process_message: Manager received command 'Command' [Mar 10 12:07:03] DEBUG[18583]: manager.c:1973 process_message: Manager received command 'Command' [Mar 10 12:07:04] DEBUG[18583]: manager.c:1973 process_message: Manager received command 'Logoff' [Mar 10 12:07:04] == Manager 'MessageNet' logged off from 127.0.0.1 set verbose 4 Verbosity was 5 and is now 4 *CLI> core[Mar 10 12:07:10] DEBUG[18555]: chan_sip.c:2008 __sip_autodestruct: Auto destroying SIP dialog '1319586744@192.168.1.97' [Mar 10 12:07:10] DEBUG[18555]: chan_sip.c:3107 sip_destroy: Destroying SIP dialog 1319586744@192.168.1.97 [Mar 10 12:07:10] Really destroying SIP dialog '1319586744@192.168.1.97' Method: REGISTER [Mar 10 12:07:10] DEBUG[18555]: chan_sip.c:2008 __sip_autodestruct: Auto destroying SIP dialog '000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165' [Mar 10 12:07:10] DEBUG[18555]: chan_sip.c:3107 sip_destroy: Destroying SIP dialog 000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165 [Mar 10 12:07:10] Really destroying SIP dialog '000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165' Method: REGISTER [Mar 10 12:07:10] DEBUG[18555]: chan_sip.c:2008 __sip_autodestruct: Auto destroying SIP dialog '55ae6647666e@24.123.23.170' [Mar 10 12:07:10] DEBUG[18555]: chan_sip.c:3107 sip_destroy: Destroying SIP dialog 55ae6647666e@24.123.23.170 [Mar 10 12:07:10] Really destroying SIP dialog '55ae6647666e@24.123.23.170' Method: REGISTER [Mar 10 12:07:10] DEBUG[18555]: chan_sip.c:2008 __sip_autodestruct: Auto destroying SIP dialog '55ae66469f11@24.123.23.170' [Mar 10 12:07:10] DEBUG[18555]: chan_sip.c:3107 sip_destroy: Destroying SIP dialog 55ae66469f11@24.123.23.170 [Mar 10 12:07:10] Really destroying SIP dialog '55ae66469f11@24.123.23.170' Method: REGISTER [Mar 10 12:07:10] DEBUG[18555]: chan_sip.c:2008 __sip_autodestruct: Auto destroying SIP dialog '55ae66476793@24.123.23.170' [Mar 10 12:07:10] DEBUG[18555]: chan_sip.c:3107 sip_destroy: Destroying SIP dialog 55ae66476793@24.123.23.170 [Mar 10 12:07:10] Really destroying SIP dialog '55ae66476793@24.123.23.170' Method: REGISTER [Mar 10 12:07:10] DEBUG[18555]: chan_sip.c:2008 __sip_autodestruct: Auto destroying SIP dialog '55ae6647678c@24.123.23.170' [Mar 10 12:07:10] DEBUG[18555]: chan_sip.c:3107 sip_destroy: Destroying SIP dialog 55ae6647678c@24.123.23.170 [Mar 10 12:07:10] Really destroying SIP dialog '55ae6647678c@24.123.23.170' Method: REGISTER [Mar 10 12:07:11] DEBUG[18555]: chan_sip.c:2008 __sip_autodestruct: Auto destroying SIP dialog '000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165' [Mar 10 12:07:11] DEBUG[18555]: chan_sip.c:3107 sip_destroy: Destroying SIP dialog 000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165 [Mar 10 12:07:11] Really destroying SIP dialog '000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165' Method: REGISTER [Mar 10 12:07:12] DEBUG[18555]: chan_sip.c:2008 __sip_autodestruct: Auto destroying SIP dialog '55ae664767e0@24.123.23.170' [Mar 10 12:07:12] DEBUG[18555]: chan_sip.c:3107 sip_destroy: Destroying SIP dialog 55ae664767e0@24.123.23.170 [Mar 10 12:07:12] Really destroying SIP dialog '55ae664767e0@24.123.23.170' Method: REGISTER sip debug SIP Debugging enabled The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead. *CLI> [Mar 10 12:07:14] DEBUG[18555]: chan_sip.c:2008 __sip_autodestruct: Auto destroying SIP dialog '000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165' [Mar 10 12:07:14] DEBUG[18555]: chan_sip.c:3107 sip_destroy: Destroying SIP dialog 000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165 [Mar 10 12:07:14] Really destroying SIP dialog '000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165' Method: REGISTER [Mar 10 12:07:16] DEBUG[18555]: chan_sip.c:2008 __sip_autodestruct: Auto destroying SIP dialog '000ff78d-ebb20007-4ad25281-15520cf9@192.168.1.165' [Mar 10 12:07:16] DEBUG[18555]: chan_sip.c:3107 sip_destroy: Destroying SIP dialog 000ff78d-ebb20007-4ad25281-15520cf9@192.168.1.165 [Mar 10 12:07:16] Really destroying SIP dialog '000ff78d-ebb20007-4ad25281-15520cf9@192.168.1.165' Method: REGISTER [Mar 10 12:07:18] DEBUG[18543]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel IAX2/606 [Mar 10 12:07:18] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for IAX2 - 606 [Mar 10 12:07:18] DEBUG[18527]: chan_iax2.c:9636 iax2_devicestate: Checking device state for device 606 [Mar 10 12:07:18] DEBUG[18527]: chan_iax2.c:9644 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 606? addr=-1509840704, defaddr=0 maxms=0, lastms=0 [Mar 10 12:07:18] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for IAX2/606 - state 1 (Not in use) [Mar 10 12:07:18] DEBUG[18585]: 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. sip set debug SIP Debugging re-enabled *CLI> [Mar 10 12:07:20] DEBUG[18548]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel IAX2/616 [Mar 10 12:07:20] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for IAX2 - 616 [Mar 10 12:07:20] DEBUG[18527]: chan_iax2.c:9636 iax2_devicestate: Checking device state for device 616 [Mar 10 12:07:20] DEBUG[18527]: chan_iax2.c:9644 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 616? addr=1159759178, defaddr=0 maxms=0, lastms=0 [Mar 10 12:07:20] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for IAX2/616 - state 1 (Not in use) [Mar 10 12:07:20] DEBUG[18586]: 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. [Mar 10 12:07:20] DEBUG[18555]: chan_sip.c:2008 __sip_autodestruct: Auto destroying SIP dialog '55ae66469f7f@24.123.23.170' [Mar 10 12:07:20] DEBUG[18555]: chan_sip.c:3107 sip_destroy: Destroying SIP dialog 55ae66469f7f@24.123.23.170 [Mar 10 12:07:20] Really destroying SIP dialog '55ae66469f7f@24.123.23.170' Method: REGISTER [Mar 10 12:07:21] <--- 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=z9hG4bK065576d5 From: To: Call-ID: 000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165 CSeq: 11586 REGISTER User-Agent: CSCO/7 Contact: Content-Length: 0 Expires: 60 <-------------> [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK065576d5 (58) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: From: (40) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: (38) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165 (58) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: CSeq: 11586 REGISTER (20) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: User-Agent: CSCO/7 (18) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Contact: (37) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Content-Length: 0 (17) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Expires: 60 (11) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: (0) [Mar 10 12:07:21] --- (10 headers 0 lines) --- [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4308 sip_alloc: Allocating new SIP dialog for 000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165 - REGISTER (No RTP) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:07:21] Using latest REGISTER request as basis request [Mar 10 12:07:21] Sending to 192.168.1.165 : 5060 (no NAT) [Mar 10 12:07:21] <--- 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=z9hG4bK065576d5;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165 CSeq: 11586 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:21] <--- 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=z9hG4bK065576d5;received=192.168.1.165 From: To: ;tag=as5a47b0ed Call-ID: 000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165 CSeq: 11586 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="24da9741" Content-Length: 0 <------------> [Mar 10 12:07:21] Scheduling destruction of SIP dialog '000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165' in 32000 ms (Method: REGISTER) [Mar 10 12:07:21] NOTICE[18555]: chan_sip.c:7128 sip_reregister: -- Re-registration for 3173241052@sip.broadvoice.com [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4308 sip_alloc: Allocating new SIP dialog for 2c01f4fc3cc0317e799f35d0601622e1@24.123.23.170 - REGISTER (No RTP) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:7287 transmit_register: Scheduled a registration timeout for sip.broadvoice.com id #159 [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:7334 transmit_register: >>> Re-using Auth data for 3173241052@sip.broadvoice.com [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:sip.broadvoice.com SIP/2.0 (39) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK231da64d;rport (64) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: From: ;tag=as2b15325e (56) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: (39) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 2c01f4fc3cc0317e799f35d0601622e1@24.123.23.170 (55) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: CSeq: 105 REGISTER (18) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Max-Forwards: 70 (16) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Authorization: Digest username="3173241052", realm="BroadWorks", algorithm=MD5, uri="sip:sip.broadvoice.com", nonce="BroadWorksXez4b7xcwTti1x0kBW", response="37be7040994df0d1ea2c198f38ad1f6e", opaque="", qop=auth, cnonce="74afd308", nc=00000003 (244) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Expires: 120 (12) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Contact: (39) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Event: registration (19) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Content-Length: 0 (17) [Mar 10 12:07:21] REGISTER 13 headers, 0 lines [Mar 10 12:07:21] REGISTER attempt 1 to 3173241052@sip.broadvoice.com [Mar 10 12:07:21] 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=z9hG4bK231da64d;rport From: ;tag=as2b15325e To: Call-ID: 2c01f4fc3cc0317e799f35d0601622e1@24.123.23.170 CSeq: 105 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="3173241052", realm="BroadWorks", algorithm=MD5, uri="sip:sip.broadvoice.com", nonce="BroadWorksXez4b7xcwTti1x0kBW", response="37be7040994df0d1ea2c198f38ad1f6e", opaque="", qop=auth, cnonce="74afd308", nc=00000003 Expires: 120 Contact: Event: registration Content-Length: 0 --- [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #160 [Mar 10 12:07:21] <--- SIP read from 147.135.12.128:5060 ---> SIP/2.0 200 OK Call-ID: 2c01f4fc3cc0317e799f35d0601622e1@24.123.23.170 CSeq: 105 REGISTER From: ;tag=as2b15325e To: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK231da64d Contact: Expires: 30 Event: registration Content-Length: 0 <-------------> [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: SIP/2.0 200 OK (14) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Call-ID: 2c01f4fc3cc0317e799f35d0601622e1@24.123.23.170 (55) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 105 REGISTER (18) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: From: ;tag=as2b15325e (56) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: To: (39) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK231da64d (58) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: (39) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Expires: 30 (11) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Event: registration (19) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (20) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: (0) [Mar 10 12:07:21] --- (10 headers 0 lines) --- [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #160 [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '2c01f4fc3cc0317e799f35d0601622e1@24.123.23.170' of Request 105: Match Not Found [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:12025 handle_response_register: Registration successful [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:12028 handle_response_register: Cancelling timeout 159 [Mar 10 12:07:21] Scheduling destruction of SIP dialog '2c01f4fc3cc0317e799f35d0601622e1@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:07:21] NOTICE[18555]: chan_sip.c:12080 handle_response_register: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 23 s) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:2008 __sip_autodestruct: Auto destroying SIP dialog '55ae664696c0@24.123.23.170' [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:3107 sip_destroy: Destroying SIP dialog 55ae664696c0@24.123.23.170 [Mar 10 12:07:21] Really destroying SIP dialog '55ae664696c0@24.123.23.170' Method: REGISTER [Mar 10 12:07:21] <--- 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=z9hG4bK3845a0a7 From: To: Call-ID: 000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165 CSeq: 11587 REGISTER User-Agent: CSCO/7 Contact: Authorization: Digest username="511",realm="asterisk",uri="sip:24.123.23.170",response="bae49a6c795d910cadcb63afca88a56d",nonce="24da9741",algorithm=MD5 Content-Length: 0 Expires: 60 <-------------> [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK3845a0a7 (58) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: From: (40) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: (38) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165 (58) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: CSeq: 11587 REGISTER (20) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: User-Agent: CSCO/7 (18) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Contact: (37) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Authorization: Digest username="511",realm="asterisk",uri="sip:24.123.23.170",response="bae49a6c795d910cadcb63afca88a56d",nonce="24da9741",algorithm=MD5 (152) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Expires: 60 (11) [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: (0) [Mar 10 12:07:21] --- (11 headers 0 lines) --- [Mar 10 12:07:21] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:07:21] Using latest REGISTER request as basis request [Mar 10 12:07:21] Sending to 192.168.1.165 : 5060 (no NAT) [Mar 10 12:07:21] <--- 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=z9hG4bK3845a0a7;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165 CSeq: 11587 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:21] -- Saved useragent "CSCO/7" for peer 511 [Mar 10 12:07:21] <--- 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=z9hG4bK3845a0a7;received=192.168.1.165 From: To: ;tag=as5a47b0ed Call-ID: 000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165 CSeq: 11587 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Sat, 10 Mar 2007 17:07:21 GMT Content-Length: 0 <------------> [Mar 10 12:07:21] DEBUG[18555]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/511 [Mar 10 12:07:21] Scheduling destruction of SIP dialog '000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165' in 32000 ms (Method: REGISTER) [Mar 10 12:07:21] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 511 [Mar 10 12:07:21] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 511 [Mar 10 12:07:21] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/511 - state 1 (Not in use) [Mar 10 12:07:21] DEBUG[18587]: 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. [Mar 10 12:07:21] <--- SIP read from 68.58.36.157:5060 ---> <-------------> [Mar 10 12:07:21] --- (0 headers 0 lines) Nat keepalive --- [Mar 10 12:07:21] <--- SIP read from 192.168.1.86:5062 ---> <-------------> [Mar 10 12:07:21] --- (0 headers 0 lines) Nat keepalive --- [Mar 10 12:07:22] DEBUG[18588]: manager.c:1973 process_message: Manager received command 'Login' [Mar 10 12:07:22] == Parsing '/etc/asterisk/manager.conf': [Mar 10 12:07:22] DEBUG[18588]: config.c:851 config_text_file_load: Parsing /etc/asterisk/manager.conf [Mar 10 12:07:22] Found [Mar 10 12:07:22] DEBUG[18588]: acl.c:200 ast_append_ha: 127.0.0.1/255.255.255.0/255.255.255.0 appended to acl for peer [Mar 10 12:07:22] DEBUG[18588]: acl.c:215 ast_apply_ha: ##### Testing 127.0.0.1 with 127.0.0.0 [Mar 10 12:07:22] == Manager 'MessageNet' logged on from 127.0.0.1 [Mar 10 12:07:22] DEBUG[18588]: manager.c:1973 process_message: Manager received command 'Command' [Mar 10 12:07:22] DEBUG[18588]: manager.c:1973 process_message: Manager received command 'Command' [Mar 10 12:07:23] DEBUG[18588]: manager.c:1973 process_message: Manager received command 'Logoff' [Mar 10 12:07:23] == Manager 'MessageNet' logged off from 127.0.0.1 [Mar 10 12:07:24] <--- 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=z9hG4bK24ef5aa0 From: To: Call-ID: 000ff78d-ebb20002-23a7f663-5bd85286@192.168.1.165 CSeq: 11599 REGISTER User-Agent: CSCO/7 Contact: Content-Length: 0 Expires: 60 <-------------> [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK24ef5aa0 (58) [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: From: (40) [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: (38) [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 000ff78d-ebb20002-23a7f663-5bd85286@192.168.1.165 (58) [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: CSeq: 11599 REGISTER (20) [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: User-Agent: CSCO/7 (18) [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Contact: (37) [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Content-Length: 0 (17) [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Expires: 60 (11) [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: (0) [Mar 10 12:07:24] --- (10 headers 0 lines) --- [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:4308 sip_alloc: Allocating new SIP dialog for 000ff78d-ebb20002-23a7f663-5bd85286@192.168.1.165 - REGISTER (No RTP) [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:07:24] Using latest REGISTER request as basis request [Mar 10 12:07:24] Sending to 192.168.1.165 : 5060 (no NAT) [Mar 10 12:07:24] <--- 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=z9hG4bK24ef5aa0;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20002-23a7f663-5bd85286@192.168.1.165 CSeq: 11599 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:24] <--- 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=z9hG4bK24ef5aa0;received=192.168.1.165 From: To: ;tag=as674e5c80 Call-ID: 000ff78d-ebb20002-23a7f663-5bd85286@192.168.1.165 CSeq: 11599 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6583ca8b" Content-Length: 0 <------------> [Mar 10 12:07:24] Scheduling destruction of SIP dialog '000ff78d-ebb20002-23a7f663-5bd85286@192.168.1.165' in 32000 ms (Method: REGISTER) [Mar 10 12:07:24] <--- 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=z9hG4bK354fb1eb From: To: Call-ID: 000ff78d-ebb20002-23a7f663-5bd85286@192.168.1.165 CSeq: 11600 REGISTER User-Agent: CSCO/7 Contact: Authorization: Digest username="510",realm="asterisk",uri="sip:24.123.23.170",response="97092b5b41b10aa9839af198969ba82b",nonce="6583ca8b",algorithm=MD5 Content-Length: 0 Expires: 60 <-------------> [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK354fb1eb (58) [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: From: (40) [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: (38) [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 000ff78d-ebb20002-23a7f663-5bd85286@192.168.1.165 (58) [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: CSeq: 11600 REGISTER (20) [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: User-Agent: CSCO/7 (18) [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Contact: (37) [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Authorization: Digest username="510",realm="asterisk",uri="sip:24.123.23.170",response="97092b5b41b10aa9839af198969ba82b",nonce="6583ca8b",algorithm=MD5 (152) [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Expires: 60 (11) [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: (0) [Mar 10 12:07:24] --- (11 headers 0 lines) --- [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:07:24] Using latest REGISTER request as basis request [Mar 10 12:07:24] Sending to 192.168.1.165 : 5060 (no NAT) [Mar 10 12:07:24] <--- 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=z9hG4bK354fb1eb;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20002-23a7f663-5bd85286@192.168.1.165 CSeq: 11600 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:24] -- Saved useragent "CSCO/7" for peer 510 [Mar 10 12:07:24] <--- 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=z9hG4bK354fb1eb;received=192.168.1.165 From: To: ;tag=as674e5c80 Call-ID: 000ff78d-ebb20002-23a7f663-5bd85286@192.168.1.165 CSeq: 11600 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Sat, 10 Mar 2007 17:07:24 GMT Content-Length: 0 <------------> [Mar 10 12:07:24] DEBUG[18555]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/510 [Mar 10 12:07:24] Scheduling destruction of SIP dialog '000ff78d-ebb20002-23a7f663-5bd85286@192.168.1.165' in 32000 ms (Method: REGISTER) [Mar 10 12:07:24] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 510 [Mar 10 12:07:24] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 510 [Mar 10 12:07:24] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/510 - state 1 (Not in use) [Mar 10 12:07:24] DEBUG[18589]: 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. [Mar 10 12:07:24] <--- SIP read from 192.168.1.97:5060 ---> REGISTER sip:24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.97:5060;rport;branch=z9hG4bK3984333619 From: "Display Name" ;tag=1786759218 To: "Display Name" Call-ID: 1319586744@192.168.1.97 CSeq: 62 REGISTER Contact: ;action=proxy max-forwards: 70 expires: 60 user-agent: UTSTARCOM F1000/Device ID-0007ba26174b Content-Length: 0 <-------------> [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.97:5060;rport;branch=z9hG4bK3984333619 (65) [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: From: "Display Name" ;tag=1786759218 (59) [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: "Display Name" (42) [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 1319586744@192.168.1.97 (32) [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: CSeq: 62 REGISTER (17) [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;action=proxy (49) [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: max-forwards: 70 (16) [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: expires: 60 (11) [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: user-agent: UTSTARCOM F1000/Device ID-0007ba26174b (50) [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Content-Length: 0 (17) [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: (0) [Mar 10 12:07:24] --- (11 headers 0 lines) --- [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:4308 sip_alloc: Allocating new SIP dialog for 1319586744@192.168.1.97 - REGISTER (No RTP) [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:07:24] Using latest REGISTER request as basis request [Mar 10 12:07:24] Sending to 192.168.1.97 : 5060 (NAT) [Mar 10 12:07:24] <--- Transmitting (NAT) to 192.168.1.97:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.97:5060;branch=z9hG4bK3984333619;received=192.168.1.97;rport=5060 From: "Display Name" ;tag=1786759218 To: "Display Name" Call-ID: 1319586744@192.168.1.97 CSeq: 62 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:24] <--- Transmitting (NAT) to 192.168.1.97:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.97:5060;branch=z9hG4bK3984333619;received=192.168.1.97;rport=5060 From: "Display Name" ;tag=1786759218 To: "Display Name" ;tag=as2f6cb7e2 Call-ID: 1319586744@192.168.1.97 CSeq: 62 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="67fb71d3" Content-Length: 0 <------------> [Mar 10 12:07:24] Scheduling destruction of SIP dialog '1319586744@192.168.1.97' in 32000 ms (Method: REGISTER) [Mar 10 12:07:24] <--- SIP read from 192.168.1.97:5060 ---> <-------------> [Mar 10 12:07:24] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: (0) [Mar 10 12:07:24] --- (0 headers 0 lines) Nat keepalive --- [Mar 10 12:07:25] <--- SIP read from 192.168.1.97:5060 ---> REGISTER sip:24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.97:5060;rport;branch=z9hG4bK3984333619 From: "Display Name" ;tag=1786759218 To: "Display Name" Call-ID: 1319586744@192.168.1.97 CSeq: 62 REGISTER Contact: ;action=proxy max-forwards: 70 expires: 60 user-agent: UTSTARCOM F1000/Device ID-0007ba26174b Content-Length: 0 <-------------> [Mar 10 12:07:25] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:07:25] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.97:5060;rport;branch=z9hG4bK3984333619 (65) [Mar 10 12:07:25] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: From: "Display Name" ;tag=1786759218 (59) [Mar 10 12:07:25] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: "Display Name" (42) [Mar 10 12:07:25] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 1319586744@192.168.1.97 (32) [Mar 10 12:07:25] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: CSeq: 62 REGISTER (17) [Mar 10 12:07:25] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;action=proxy (49) [Mar 10 12:07:25] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: max-forwards: 70 (16) [Mar 10 12:07:25] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: expires: 60 (11) [Mar 10 12:07:25] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: user-agent: UTSTARCOM F1000/Device ID-0007ba26174b (50) [Mar 10 12:07:25] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Content-Length: 0 (17) [Mar 10 12:07:25] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: (0) [Mar 10 12:07:25] --- (11 headers 0 lines) --- [Mar 10 12:07:25] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:07:25] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 62, ours 62) [Mar 10 12:07:25] Using latest REGISTER request as basis request [Mar 10 12:07:25] Sending to 192.168.1.97 : 5060 (NAT) [Mar 10 12:07:25] <--- Transmitting (NAT) to 192.168.1.97:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.97:5060;branch=z9hG4bK3984333619;received=192.168.1.97;rport=5060 From: "Display Name" ;tag=1786759218 To: "Display Name" Call-ID: 1319586744@192.168.1.97 CSeq: 62 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:25] <--- Transmitting (NAT) to 192.168.1.97:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.97:5060;branch=z9hG4bK3984333619;received=192.168.1.97;rport=5060 From: "Display Name" ;tag=1786759218 To: "Display Name" ;tag=as2f6cb7e2 Call-ID: 1319586744@192.168.1.97 CSeq: 62 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="67fb71d3" Content-Length: 0 <------------> [Mar 10 12:07:25] Scheduling destruction of SIP dialog '1319586744@192.168.1.97' in 32000 ms (Method: REGISTER) [Mar 10 12:07:25] <--- SIP read from 192.168.1.97:5060 ---> REGISTER sip:24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.97:5060;rport;branch=z9hG4bK1469679949 From: "Display Name" ;tag=1104220028 To: "Display Name" Call-ID: 1319586744@192.168.1.97 CSeq: 63 REGISTER Contact: ;action=proxy Authorization: Digest username="551", realm="asterisk", nonce="67fb71d3", uri="sip:24.123.23.170", response="c36df6e8b148d2184767411bc9886c20", algorithm=MD5 max-forwards: 70 expires: 60 user-agent: UTSTARCOM F1000/Device ID-0007ba26174b Content-Length: 0 <-------------> [Mar 10 12:07:25] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:07:25] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.97:5060;rport;branch=z9hG4bK1469679949 (65) [Mar 10 12:07:25] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: From: "Display Name" ;tag=1104220028 (59) [Mar 10 12:07:25] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: "Display Name" (42) [Mar 10 12:07:25] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 1319586744@192.168.1.97 (32) [Mar 10 12:07:25] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: CSeq: 63 REGISTER (17) [Mar 10 12:07:25] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;action=proxy (49) [Mar 10 12:07:25] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Authorization: Digest username="551", realm="asterisk", nonce="67fb71d3", uri="sip:24.123.23.170", response="c36df6e8b148d2184767411bc9886c20", algorithm=MD5 (157) [Mar 10 12:07:25] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: max-forwards: 70 (16) [Mar 10 12:07:25] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: expires: 60 (11) [Mar 10 12:07:25] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: user-agent: UTSTARCOM F1000/Device ID-0007ba26174b (50) [Mar 10 12:07:25] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Content-Length: 0 (17) [Mar 10 12:07:25] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: (0) [Mar 10 12:07:25] --- (12 headers 0 lines) --- [Mar 10 12:07:25] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:07:25] Using latest REGISTER request as basis request [Mar 10 12:07:25] Sending to 192.168.1.97 : 5060 (NAT) [Mar 10 12:07:25] <--- Transmitting (NAT) to 192.168.1.97:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.97:5060;branch=z9hG4bK1469679949;received=192.168.1.97;rport=5060 From: "Display Name" ;tag=1104220028 To: "Display Name" Call-ID: 1319586744@192.168.1.97 CSeq: 63 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:25] <--- Transmitting (NAT) to 192.168.1.97:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.97:5060;branch=z9hG4bK1469679949;received=192.168.1.97;rport=5060 From: "Display Name" ;tag=1104220028 To: "Display Name" ;tag=as2f6cb7e2 Call-ID: 1319586744@192.168.1.97 CSeq: 63 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Sat, 10 Mar 2007 17:07:25 GMT Content-Length: 0 <------------> [Mar 10 12:07:25] DEBUG[18555]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/551 [Mar 10 12:07:25] Scheduling destruction of SIP dialog '1319586744@192.168.1.97' in 32000 ms (Method: REGISTER) [Mar 10 12:07:25] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 551 [Mar 10 12:07:25] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 551 [Mar 10 12:07:25] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/551 - state 1 (Not in use) [Mar 10 12:07:25] DEBUG[18590]: app_queue.c:546 changethread: Device 'SIP/551' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Mar 10 12:07:26] <--- SIP read from 192.168.1.83:5060 ---> INVITE sip:523@24.123.23.170:5060 SIP/2.0 Max-Forwards: 70 Content-Length: 292 Via: SIP/2.0/UDP 192.168.1.83:5060;branch=z9hG4bK625a53b4e Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 From: 593 ;tag=48204f4e1c15803 To: 523 CSeq: 2124963504 INVITE Supported: timer Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE Content-Type: application/sdp Contact: 593 Supported: replaces User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 v=0 o=MxSIP 0 1143068287 IN IP4 192.168.1.83 s=SIP Call c=IN IP4 192.168.1.83 t=0 0 m=audio 51000 RTP/AVP 0 18 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:on - - - - <-------------> [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: INVITE sip:523@24.123.23.170:5060 SIP/2.0 (41) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Max-Forwards: 70 (16) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Content-Length: 292 (19) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Via: SIP/2.0/UDP 192.168.1.83:5060;branch=z9hG4bK625a53b4e (58) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 (54) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: From: 593 ;tag=48204f4e1c15803 (58) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: To: 523 (36) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: CSeq: 2124963504 INVITE (23) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Supported: timer (16) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE (55) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Content-Type: application/sdp (29) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Contact: 593 (40) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Supported: replaces (19) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 (42) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 14: (0) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: o=MxSIP 0 1143068287 IN IP4 192.168.1.83 (40) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: s=SIP Call (10) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: c=IN IP4 192.168.1.83 (21) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: m=audio 51000 RTP/AVP 0 18 8 101 (32) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-15 (15) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=sendrecv (10) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=silenceSupp:on - - - - (24) [Mar 10 12:07:26] --- (14 headers 14 lines) --- [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:2573 do_setnat: Setting NAT on RTP to Off [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:2578 do_setnat: Setting NAT on VRTP to Off [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4308 sip_alloc: Allocating new SIP dialog for a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 - INVITE (With RTP) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:1678 parse_sip_options: Begin: parsing SIP "Supported: timer" [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:1686 parse_sip_options: Found SIP option: -timer- [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:1692 parse_sip_options: Matched SIP option: timer [Mar 10 12:07:26] Sending to 192.168.1.83 : 5060 (no NAT) [Mar 10 12:07:26] Using INVITE request as basis request - a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:2573 do_setnat: Setting NAT on RTP to Off [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:2578 do_setnat: Setting NAT on VRTP to Off [Mar 10 12:07:26] <--- Reliably Transmitting (no NAT) to 192.168.1.83:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.83:5060;branch=z9hG4bK625a53b4e;received=192.168.1.83 From: 593 ;tag=48204f4e1c15803 To: 523 ;tag=as001f3e37 Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 CSeq: 2124963504 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5f611baa" Content-Length: 0 <------------> [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #172 [Mar 10 12:07:26] Scheduling destruction of SIP dialog 'a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83' in 32000 ms (Method: INVITE) [Mar 10 12:07:26] Found user '593' [Mar 10 12:07:26] <--- SIP read from 192.168.1.83:5060 ---> ACK sip:523@24.123.23.170:5060 SIP/2.0 Max-Forwards: 70 Content-Length: 0 Via: SIP/2.0/UDP 192.168.1.83:5060;branch=z9hG4bK625a53b4e Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 From: 593 ;tag=48204f4e1c15803 To: 523 ;tag=as001f3e37 CSeq: 2124963504 ACK User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 <-------------> [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: ACK sip:523@24.123.23.170:5060 SIP/2.0 (38) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Max-Forwards: 70 (16) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Content-Length: 0 (17) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Via: SIP/2.0/UDP 192.168.1.83:5060;branch=z9hG4bK625a53b4e (58) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 (54) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: From: 593 ;tag=48204f4e1c15803 (58) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: To: 523 ;tag=as001f3e37 (51) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: CSeq: 2124963504 ACK (20) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 (42) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: (0) [Mar 10 12:07:26] --- (9 headers 0 lines) --- [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received ACK (6) - Command in SIP ACK [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #172 [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on 'a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83' of Response 2124963504: Match Not Found [Mar 10 12:07:26] <--- SIP read from 192.168.1.83:5060 ---> INVITE sip:523@24.123.23.170:5060 SIP/2.0 Max-Forwards: 70 Content-Length: 292 Via: SIP/2.0/UDP 192.168.1.83:5060;branch=z9hG4bKe471c4bef Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 From: 593 ;tag=48204f4e1c15803 To: 523 CSeq: 2124963505 INVITE Supported: timer Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE Content-Type: application/sdp Supported: replaces Proxy-Authorization:Digest response="018693911625a090eb17a7954cfc0699",username="593",realm="asterisk",nonce="5f611baa",algorithm=MD5,uri="sip:523@24.123.23.170:5060" Contact: 593 User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 v=0 o=MxSIP 0 1143068287 IN IP4 192.168.1.83 s=SIP Call c=IN IP4 192.168.1.83 t=0 0 m=audio 51000 RTP/AVP 0 18 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:on - - - - <-------------> [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: INVITE sip:523@24.123.23.170:5060 SIP/2.0 (41) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Max-Forwards: 70 (16) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Content-Length: 292 (19) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Via: SIP/2.0/UDP 192.168.1.83:5060;branch=z9hG4bKe471c4bef (58) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 (54) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: From: 593 ;tag=48204f4e1c15803 (58) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: To: 523 (36) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: CSeq: 2124963505 INVITE (23) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Supported: timer (16) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE (55) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Content-Type: application/sdp (29) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Supported: replaces (19) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Proxy-Authorization:Digest response="018693911625a090eb17a7954cfc0699",username="593",realm="asterisk",nonce="5f611baa",algorithm=MD5,uri="sip:523@24.123.23.170:5060" (166) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: Contact: 593 (40) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 14: User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 (42) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 15: (0) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: o=MxSIP 0 1143068287 IN IP4 192.168.1.83 (40) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: s=SIP Call (10) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: c=IN IP4 192.168.1.83 (21) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: m=audio 51000 RTP/AVP 0 18 8 101 (32) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-15 (15) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=sendrecv (10) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=silenceSupp:on - - - - (24) [Mar 10 12:07:26] --- (15 headers 14 lines) --- [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Mar 10 12:07:26] Sending to 192.168.1.83 : 5060 (no NAT) [Mar 10 12:07:26] Using INVITE request as basis request - a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:2573 do_setnat: Setting NAT on RTP to Off [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:2578 do_setnat: Setting NAT on VRTP to Off [Mar 10 12:07:26] Found user '593' [Mar 10 12:07:26] Found RTP audio format 0 [Mar 10 12:07:26] Found RTP audio format 18 [Mar 10 12:07:26] Found RTP audio format 8 [Mar 10 12:07:26] Found RTP audio format 101 [Mar 10 12:07:26] Peer audio RTP is at port 192.168.1.83:51000 [Mar 10 12:07:26] Found description format PCMU for ID 0 [Mar 10 12:07:26] Found description format G729 for ID 18 [Mar 10 12:07:26] Found description format PCMA for ID 8 [Mar 10 12:07:26] Found description format telephone-event for ID 101 [Mar 10 12:07:26] Got unsupported a:fmtp in SDP offer [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:5123 process_sdp: T38 state changed to 0 on channel [Mar 10 12:07:26] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) [Mar 10 12:07:26] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 10 12:07:26] Peer audio RTP is at port 192.168.1.83:51000 [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:5203 process_sdp: We're settling with these formats: 0xc (ulaw|alaw) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:13378 handle_request_invite: Checking SIP call limits for device 593 [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:3001 update_call_counter: Updating call counter for incoming call [Mar 10 12:07:26] Looking for 523 in smvoice-sip (domain 24.123.23.170) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:3803 sip_new: *** Our native formats are 0x4 (ulaw) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:3804 sip_new: *** Joint capabilities are 0xc (ulaw|alaw) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:3805 sip_new: *** Our capabilities are 0xe (gsm|ulaw|alaw) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:3806 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:3829 sip_new: This channel will not be able to handle video. [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:7964 build_route: build_route: Contact hop: 593 [Mar 10 12:07:26] list_route: hop: [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:13453 handle_request_invite: SIP/593-b77052f8: New call is still down.... Trying... [Mar 10 12:07:26] <--- Transmitting (no NAT) to 192.168.1.83:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.83:5060;branch=z9hG4bKe471c4bef;received=192.168.1.83 From: 593 ;tag=48204f4e1c15803 To: 523 Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 CSeq: 2124963505 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:26] DEBUG[18555]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/593-b77052f8 [Mar 10 12:07:26] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 593 [Mar 10 12:07:26] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 593 [Mar 10 12:07:26] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/593 - state 1 (Not in use) [Mar 10 12:07:26] DEBUG[18591]: pbx.c:1791 pbx_extension_helper: Launching 'NoOp' [Mar 10 12:07:26] -- Executing [523@smvoice-sip:1] NoOp("SIP/593-b77052f8", "5xx") in new stack [Mar 10 12:07:26] DEBUG[18591]: pbx.c:1791 pbx_extension_helper: Launching 'Set' [Mar 10 12:07:26] -- Executing [523@smvoice-sip:2] Set("SIP/593-b77052f8", "SMVOICE_CONTEXT_EXTEN=523") in new stack [Mar 10 12:07:26] DEBUG[18591]: pbx.c:1791 pbx_extension_helper: Launching 'AGI' [Mar 10 12:07:26] -- Executing [523@smvoice-sip:3] AGI("SIP/593-b77052f8", "smvoice|-company|demonstration|-digium_asterisk|-asterisk_callat_forwarding|523") in new stack [Mar 10 12:07:26] DEBUG[18592]: app_queue.c:546 changethread: Device 'SIP/593' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Mar 10 12:07:26] -- Launched AGI Script /var/lib/asterisk/agi-bin/smvoice [Mar 10 12:07:26] -- AGI Script smvoice completed, returning 0 [Mar 10 12:07:26] DEBUG[18591]: pbx.c:1712 pbx_substitute_variables_helper_full: Expression result is '0' [Mar 10 12:07:26] DEBUG[18591]: pbx.c:1791 pbx_extension_helper: Launching 'GotoIf' [Mar 10 12:07:26] -- Executing [523@smvoice-sip:4] GotoIf("SIP/593-b77052f8", "0?INVALID|1") in new stack [Mar 10 12:07:26] DEBUG[18591]: pbx.c:5969 pbx_builtin_gotoif: Not taking any branch [Mar 10 12:07:26] DEBUG[18591]: pbx.c:1712 pbx_substitute_variables_helper_full: Expression result is '0' [Mar 10 12:07:26] DEBUG[18591]: pbx.c:1791 pbx_extension_helper: Launching 'GotoIf' [Mar 10 12:07:26] -- Executing [523@smvoice-sip:5] GotoIf("SIP/593-b77052f8", "0?_5XX-NOANSWER|1") in new stack [Mar 10 12:07:26] DEBUG[18591]: pbx.c:5969 pbx_builtin_gotoif: Not taking any branch [Mar 10 12:07:26] DEBUG[18591]: pbx.c:1791 pbx_extension_helper: Launching 'Dial' [Mar 10 12:07:26] -- Executing [523@smvoice-sip:6] Dial("SIP/593-b77052f8", "SIP/523|20|") in new stack [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:15267 sip_request_call: Asked to create a SIP channel with formats: 0x4 (ulaw) [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:4308 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:2573 do_setnat: Setting NAT on RTP to Off [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:3803 sip_new: *** Our native formats are 0x4 (ulaw) [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:3804 sip_new: *** Joint capabilities are 0x0 (nothing) [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:3805 sip_new: *** Our capabilities are 0xe (gsm|ulaw|alaw) [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:3806 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:3808 sip_new: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:3829 sip_new: This channel will not be able to handle video. [Mar 10 12:07:26] DEBUG[18591]: rtp.c:1579 ast_rtp_make_compatible: Seeded SDP of 'SIP/523-087dd9a8' with that of 'SIP/593-b77052f8' [Mar 10 12:07:26] DEBUG[18591]: channel.c:3301 ast_channel_inherit_variables: Not copying variable STACK-smvoice-sip-523-6. [Mar 10 12:07:26] DEBUG[18591]: channel.c:3301 ast_channel_inherit_variables: Not copying variable STACK-smvoice-sip-523-5. [Mar 10 12:07:26] DEBUG[18591]: channel.c:3301 ast_channel_inherit_variables: Not copying variable STACK-smvoice-sip-523-4. [Mar 10 12:07:26] DEBUG[18591]: channel.c:3301 ast_channel_inherit_variables: Not copying variable AGISTATUS. [Mar 10 12:07:26] DEBUG[18591]: channel.c:3301 ast_channel_inherit_variables: Not copying variable SMVOICE_WORK_EXTENSION. [Mar 10 12:07:26] DEBUG[18591]: channel.c:3301 ast_channel_inherit_variables: Not copying variable SMVOICE_EXTENSION. [Mar 10 12:07:26] DEBUG[18591]: channel.c:3301 ast_channel_inherit_variables: Not copying variable SMVOICE_PIN. [Mar 10 12:07:26] DEBUG[18591]: channel.c:3301 ast_channel_inherit_variables: Not copying variable SMVOICE_CALLAT. [Mar 10 12:07:26] DEBUG[18591]: channel.c:3301 ast_channel_inherit_variables: Not copying variable STACK-smvoice-sip-523-3. [Mar 10 12:07:26] DEBUG[18591]: channel.c:3301 ast_channel_inherit_variables: Not copying variable SMVOICE_CONTEXT_EXTEN. [Mar 10 12:07:26] DEBUG[18591]: channel.c:3301 ast_channel_inherit_variables: Not copying variable STACK-smvoice-sip-523-2. [Mar 10 12:07:26] DEBUG[18591]: channel.c:3301 ast_channel_inherit_variables: Not copying variable STACK-smvoice-sip-523-1. [Mar 10 12:07:26] DEBUG[18591]: channel.c:3301 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Mar 10 12:07:26] DEBUG[18591]: channel.c:3301 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT. [Mar 10 12:07:26] DEBUG[18591]: channel.c:3301 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Mar 10 12:07:26] DEBUG[18591]: channel.c:3301 ast_channel_inherit_variables: Not copying variable SIPURI. [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:2828 sip_call: Outgoing Call for 523 [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:3001 update_call_counter: Updating call counter for outgoing call [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:2843 sip_call: Our T38 capability (0), joint T38 capability (0) [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:6182 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: False [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:6183 add_sdp: ** Our prefcodec: 0x4 (ulaw) [Mar 10 12:07:26] Audio is at 24.123.23.170 port 10848 [Mar 10 12:07:26] Adding codec 0x4 (ulaw) to SDP [Mar 10 12:07:26] Adding codec 0x8 (alaw) to SDP [Mar 10 12:07:26] Adding codec 0x2 (gsm) to SDP [Mar 10 12:07:26] Adding non-codec 0x1 (telephone-event) to SDP [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:6314 add_sdp: -- Done with adding codecs to SDP [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:6359 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 0: INVITE sip:523@192.168.1.85:5060 SIP/2.0 (40) [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK13ea82fa (58) [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 2: From: "593 593" ;tag=as628c2367 (54) [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 3: To: (31) [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 4: Contact: (32) [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 5: Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 (55) [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 6: CSeq: 102 INVITE (16) [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 9: Date: Sat, 10 Mar 2007 17:07:26 GMT (35) [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 11: Supported: replaces (19) [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 12: Content-Type: application/sdp (29) [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 13: Content-Length: 289 (19) [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 14: (0) [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: o=root 18522 18522 IN IP4 24.123.23.170 (39) [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: s=session (9) [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: c=IN IP4 24.123.23.170 (22) [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: m=audio 10848 RTP/AVP 0 8 3 101 (31) [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-16 (15) [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=silenceSupp:off - - - - (25) [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=sendrecv (10) [Mar 10 12:07:26] Reliably Transmitting (no NAT) to 192.168.1.85:5060: INVITE sip:523@192.168.1.85:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK13ea82fa From: "593 593" ;tag=as628c2367 To: Contact: Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sat, 10 Mar 2007 17:07:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 289 v=0 o=root 18522 18522 IN IP4 24.123.23.170 s=session c=IN IP4 24.123.23.170 t=0 0 m=audio 10848 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 --- [Mar 10 12:07:26] DEBUG[18591]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #174 [Mar 10 12:07:26] -- Called 523 [Mar 10 12:07:26] <--- SIP read from 192.168.1.85:5060 ---> SIP/2.0 100 Trying To: ;tag=a0b2f919869f2cd75f7879cc9d8200ec Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK13ea82fa From: "593 593" ;tag=as628c2367 Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 102 INVITE Content-Length: 0 Supported: sip-cc, sip-cc-01, replaces, timer Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK <-------------> [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: SIP/2.0 100 Trying (18) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: To: ;tag=a0b2f919869f2cd75f7879cc9d8200ec (68) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK13ea82fa (58) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: From: "593 593" ;tag=as628c2367 (54) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 (55) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: CSeq: 102 INVITE (16) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Content-Length: 0 (17) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: (0) [Mar 10 12:07:26] --- (9 headers 0 lines) --- [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:2120 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #174 - INVITE (got response) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:2129 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '53c067b5491f609931bc7a0965b73fd3@24.123.23.170' Request 102: Found [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:11621 handle_response_invite: SIP response 100 to standard invite [Mar 10 12:07:26] <--- SIP read from 192.168.1.85:5060 ---> SIP/2.0 180 Ringing To: ;tag=a0b2f919869f2cd75f7879cc9d8200ec Contact: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK13ea82fa From: "593 593" ;tag=as628c2367 Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 102 INVITE Content-Length: 0 Supported: sip-cc, sip-cc-01, replaces, timer Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK <-------------> [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: SIP/2.0 180 Ringing (19) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: To: ;tag=a0b2f919869f2cd75f7879cc9d8200ec (68) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Contact: (36) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK13ea82fa (58) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: "593 593" ;tag=as628c2367 (54) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 (55) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: CSeq: 102 INVITE (16) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Content-Length: 0 (17) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: (0) [Mar 10 12:07:26] --- (10 headers 0 lines) --- [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:2129 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '53c067b5491f609931bc7a0965b73fd3@24.123.23.170' Request 102: Found [Mar 10 12:07:26] DEBUG[18555]: chan_sip.c:11621 handle_response_invite: SIP response 180 to standard invite [Mar 10 12:07:26] DEBUG[18555]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/523-087dd9a8 [Mar 10 12:07:26] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 523 [Mar 10 12:07:26] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 523 [Mar 10 12:07:26] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/523 - state 1 (Not in use) [Mar 10 12:07:26] -- SIP/523-087dd9a8 is ringing [Mar 10 12:07:26] DEBUG[18591]: rtp.c:1514 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/593-b77052f8' with that of 'SIP/523-087dd9a8' [Mar 10 12:07:26] DEBUG[18594]: 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. [Mar 10 12:07:26] <--- Transmitting (no NAT) to 192.168.1.83:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.83:5060;branch=z9hG4bKe471c4bef;received=192.168.1.83 From: 593 ;tag=48204f4e1c15803 To: 523 ;tag=as43feeb80 Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 CSeq: 2124963505 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:2008 __sip_autodestruct: Auto destroying SIP dialog '2c01f4fc3cc0317e799f35d0601622e1@24.123.23.170' [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:3107 sip_destroy: Destroying SIP dialog 2c01f4fc3cc0317e799f35d0601622e1@24.123.23.170 [Mar 10 12:07:29] Really destroying SIP dialog '2c01f4fc3cc0317e799f35d0601622e1@24.123.23.170' Method: REGISTER [Mar 10 12:07:29] <--- SIP read from 192.168.1.85:5060 ---> SIP/2.0 200 OK To: ;tag=a0b2f919869f2cd75f7879cc9d8200ec Contact: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK13ea82fa From: "593 593" ;tag=as628c2367 Call-ID: 53c067b5491f609931bc7a0965b73fd3@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=- 18522 204404 IN IP4 192.168.1.85 s=session c=IN IP4 192.168.1.85 t=0 0 m=audio 30016 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 <-------------> [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: SIP/2.0 200 OK (14) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: To: ;tag=a0b2f919869f2cd75f7879cc9d8200ec (68) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Contact: (36) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK13ea82fa (58) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: "593 593" ;tag=as628c2367 (54) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 (55) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: CSeq: 102 INVITE (16) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Content-Type: application/sdp (29) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Content-Length: 247 (19) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: (0) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: o=- 18522 204404 IN IP4 192.168.1.85 (36) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: s=session (9) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: c=IN IP4 192.168.1.85 (21) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: m=audio 30016 RTP/AVP 0 101 (27) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-15,16,17,18 (24) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=silenceSupp:off - - - - (25) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=sendrecv (10) [Mar 10 12:07:29] --- (11 headers 12 lines) --- [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:2069 __sip_ack: Acked pending invite 102 [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '53c067b5491f609931bc7a0965b73fd3@24.123.23.170' of Request 102: Match Not Found [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:11621 handle_response_invite: SIP response 200 to standard invite [Mar 10 12:07:29] Found RTP audio format 0 [Mar 10 12:07:29] Found RTP audio format 101 [Mar 10 12:07:29] Peer audio RTP is at port 192.168.1.85:30016 [Mar 10 12:07:29] Found description format PCMU for ID 0 [Mar 10 12:07:29] Found description format telephone-event for ID 101 [Mar 10 12:07:29] Got unsupported a:fmtp in SDP offer [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:5123 process_sdp: T38 state changed to 0 on channel SIP/523-087dd9a8 [Mar 10 12:07:29] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 10 12:07:29] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 10 12:07:29] Peer audio RTP is at port 192.168.1.85:30016 [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:5203 process_sdp: We're settling with these formats: 0x4 (ulaw) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:5210 process_sdp: We have an owner, now see if we need to change this call [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:3001 update_call_counter: Updating call counter for outgoing call [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:7964 build_route: build_route: Contact hop: [Mar 10 12:07:29] list_route: hop: [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:5637 reqprep: Strict routing enforced for session 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 [Mar 10 12:07:29] set_destination: Parsing for address/port to send to [Mar 10 12:07:29] set_destination: set destination to 192.168.1.85, port 5060 [Mar 10 12:07:29] Transmitting (no NAT) to 192.168.1.85:5060: ACK sip:523@192.168.1.85:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK1c7485de From: "593 593" ;tag=as628c2367 To: ;tag=a0b2f919869f2cd75f7879cc9d8200ec Contact: Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 10 12:07:29] -- Call on SIP/523-087dd9a8 left from hold [Mar 10 12:07:29] DEBUG[18591]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/523-087dd9a8 [Mar 10 12:07:29] -- SIP/523-087dd9a8 answered SIP/593-b77052f8 [Mar 10 12:07:29] DEBUG[18591]: rtp.c:1514 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/593-b77052f8' with that of 'SIP/523-087dd9a8' [Mar 10 12:07:29] DEBUG[18591]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/593-b77052f8 [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:3461 sip_answer: SIP answering channel: SIP/593-b77052f8 [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:6414 transmit_response_with_sdp: Setting framing from config on incoming call [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:6182 add_sdp: ** Our capability: 0xc (ulaw|alaw) Video flag: True [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:6183 add_sdp: ** Our prefcodec: 0x0 (nothing) [Mar 10 12:07:29] Audio is at 24.123.23.170 port 14996 [Mar 10 12:07:29] Adding codec 0x4 (ulaw) to SDP [Mar 10 12:07:29] Adding codec 0x8 (alaw) to SDP [Mar 10 12:07:29] Adding non-codec 0x1 (telephone-event) to SDP [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:6314 add_sdp: -- Done with adding codecs to SDP [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:6359 add_sdp: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Mar 10 12:07:29] <--- Reliably Transmitting (no NAT) to 192.168.1.83:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.83:5060;branch=z9hG4bKe471c4bef;received=192.168.1.83 From: 593 ;tag=48204f4e1c15803 To: 523 ;tag=as43feeb80 Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 CSeq: 2124963505 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 266 v=0 o=root 18522 18522 IN IP4 24.123.23.170 s=session c=IN IP4 24.123.23.170 t=0 0 m=audio 14996 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 <------------> [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #176 [Mar 10 12:07:29] -- Native bridging SIP/593-b77052f8 and SIP/523-087dd9a8 [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:16865 sip_set_rtp_peer: Deferring reinvite on SIP 'a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83' - It's audio will be redirected to IP 192.168.1.85 [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:16860 sip_set_rtp_peer: Sending reinvite on SIP '53c067b5491f609931bc7a0965b73fd3@24.123.23.170' - It's audio soon redirected to IP 192.168.1.83 [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:5637 reqprep: Strict routing enforced for session 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 [Mar 10 12:07:29] set_destination: Parsing for address/port to send to [Mar 10 12:07:29] set_destination: set destination to 192.168.1.85, port 5060 [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:6182 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: True [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:6183 add_sdp: ** Our prefcodec: 0x4 (ulaw) [Mar 10 12:07:29] Audio is at 24.123.23.170 port 10848 [Mar 10 12:07:29] Adding codec 0x4 (ulaw) to SDP [Mar 10 12:07:29] Adding non-codec 0x1 (telephone-event) to SDP [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:6314 add_sdp: -- Done with adding codecs to SDP [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:6359 add_sdp: Done building SDP. Settling with this capability: 0x4 (ulaw) [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:1619 initialize_initreq: Initializing already initialized SIP dialog 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 (presumably reinvite) [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 0: INVITE sip:523@192.168.1.85:5060 SIP/2.0 (40) [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK36d806a4 (58) [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 2: From: "593 593" ;tag=as628c2367 (54) [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 3: To: ;tag=a0b2f919869f2cd75f7879cc9d8200ec (68) [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 4: Contact: (32) [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 5: Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 (55) [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 6: CSeq: 103 INVITE (16) [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Mar 10 12:07:29] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 523 [Mar 10 12:07:29] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 523 [Mar 10 12:07:29] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/523 - state 1 (Not in use) [Mar 10 12:07:29] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 593 [Mar 10 12:07:29] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 593 [Mar 10 12:07:29] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/593 - state 1 (Not in use) [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 10: Supported: replaces (19) [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 12: Content-Type: application/sdp (29) [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 13: Content-Length: 240 (19) [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 14: (0) [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: o=root 18522 18523 IN IP4 192.168.1.83 (38) [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: s=session (9) [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: c=IN IP4 192.168.1.83 (21) [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: m=audio 51000 RTP/AVP 0 101 (27) [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-16 (15) [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=silenceSupp:off - - - - (25) [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=sendrecv (10) [Mar 10 12:07:29] Reliably Transmitting (no NAT) to 192.168.1.85:5060: INVITE sip:523@192.168.1.85:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK36d806a4 From: "593 593" ;tag=as628c2367 To: ;tag=a0b2f919869f2cd75f7879cc9d8200ec Contact: Call-ID: 53c067b5491f609931bc7a0965b73fd3@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 18522 18523 IN IP4 192.168.1.83 s=session c=IN IP4 192.168.1.83 t=0 0 m=audio 51000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Mar 10 12:07:29] DEBUG[18591]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #177 [Mar 10 12:07:29] DEBUG[18595]: 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. [Mar 10 12:07:29] DEBUG[18596]: app_queue.c:546 changethread: Device 'SIP/593' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Mar 10 12:07:29] <--- SIP read from 192.168.1.83:5060 ---> ACK sip:523@24.123.23.170 SIP/2.0 Max-Forwards: 70 Content-Length: 0 Via: SIP/2.0/UDP 192.168.1.83:5060;branch=z9hG4bKe2046c113 Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 From: 593 ;tag=48204f4e1c15803 To: 523 ;tag=as43feeb80 CSeq: 2124963505 ACK Proxy-Authorization:Digest response="018693911625a090eb17a7954cfc0699",username="593",realm="asterisk",nonce="5f611baa",algorithm=MD5,uri="sip:523@24.123.23.170:5060" User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 <-------------> [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: ACK sip:523@24.123.23.170 SIP/2.0 (33) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Max-Forwards: 70 (16) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Content-Length: 0 (17) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Via: SIP/2.0/UDP 192.168.1.83:5060;branch=z9hG4bKe2046c113 (58) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 (54) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: From: 593 ;tag=48204f4e1c15803 (58) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: To: 523 ;tag=as43feeb80 (51) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: CSeq: 2124963505 ACK (20) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Proxy-Authorization:Digest response="018693911625a090eb17a7954cfc0699",username="593",realm="asterisk",nonce="5f611baa",algorithm=MD5,uri="sip:523@24.123.23.170:5060" (166) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 (42) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: (0) [Mar 10 12:07:29] --- (10 headers 0 lines) --- [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received ACK (6) - Command in SIP ACK [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #176 [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on 'a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83' of Response 2124963505: Match Not Found [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:11602 check_pendings: Sending pending reinvite on 'a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83' [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:5637 reqprep: Strict routing enforced for session a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 [Mar 10 12:07:29] set_destination: Parsing for address/port to send to [Mar 10 12:07:29] set_destination: set destination to 192.168.1.83, port 5060 [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:6182 add_sdp: ** Our capability: 0xc (ulaw|alaw) Video flag: True [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:6183 add_sdp: ** Our prefcodec: 0x0 (nothing) [Mar 10 12:07:29] Audio is at 24.123.23.170 port 14996 [Mar 10 12:07:29] Adding codec 0x4 (ulaw) to SDP [Mar 10 12:07:29] Adding codec 0x8 (alaw) to SDP [Mar 10 12:07:29] Adding non-codec 0x1 (telephone-event) to SDP [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:6314 add_sdp: -- Done with adding codecs to SDP [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:6359 add_sdp: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:1619 initialize_initreq: Initializing already initialized SIP dialog a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 (presumably reinvite) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: INVITE sip:593@192.168.1.83:5060 SIP/2.0 (40) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK150e6c3d (58) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: From: 523 ;tag=as43feeb80 (53) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: 593 ;tag=48204f4e1c15803 (56) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Contact: (32) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 (54) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: CSeq: 102 INVITE (16) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: replaces (19) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Content-Type: application/sdp (29) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: Content-Length: 264 (19) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 14: (0) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: o=root 18522 18523 IN IP4 192.168.1.85 (38) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: s=session (9) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: c=IN IP4 192.168.1.85 (21) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: m=audio 30016 RTP/AVP 0 8 101 (29) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-16 (15) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=silenceSupp:off - - - - (25) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=sendrecv (10) [Mar 10 12:07:29] Reliably Transmitting (no NAT) to 192.168.1.83:5060: INVITE sip:593@192.168.1.83:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK150e6c3d From: 523 ;tag=as43feeb80 To: 593 ;tag=48204f4e1c15803 Contact: Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 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: 264 v=0 o=root 18522 18523 IN IP4 192.168.1.85 s=session c=IN IP4 192.168.1.85 t=0 0 m=audio 30016 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 --- [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #178 [Mar 10 12:07:29] <--- SIP read from 192.168.1.85:5060 ---> SIP/2.0 200 OK To: ;tag=a0b2f919869f2cd75f7879cc9d8200ec Contact: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK36d806a4 From: "593 593" ;tag=as628c2367 Call-ID: 53c067b5491f609931bc7a0965b73fd3@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=- 18522 204405 IN IP4 192.168.1.85 s=session c=IN IP4 192.168.1.85 t=0 0 m=audio 30016 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 <-------------> [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: SIP/2.0 200 OK (14) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: To: ;tag=a0b2f919869f2cd75f7879cc9d8200ec (68) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Contact: (36) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK36d806a4 (58) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: "593 593" ;tag=as628c2367 (54) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 (55) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: CSeq: 103 INVITE (16) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Content-Type: application/sdp (29) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Content-Length: 247 (19) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: (0) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: o=- 18522 204405 IN IP4 192.168.1.85 (36) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: s=session (9) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: c=IN IP4 192.168.1.85 (21) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: m=audio 30016 RTP/AVP 0 101 (27) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-15,16,17,18 (24) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=silenceSupp:off - - - - (25) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=sendrecv (10) [Mar 10 12:07:29] --- (11 headers 12 lines) --- [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:2069 __sip_ack: Acked pending invite 103 [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #177 [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '53c067b5491f609931bc7a0965b73fd3@24.123.23.170' of Request 103: Match Not Found [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:11619 handle_response_invite: SIP response 200 to RE-invite on outgoing call 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 [Mar 10 12:07:29] Found RTP audio format 0 [Mar 10 12:07:29] Found RTP audio format 101 [Mar 10 12:07:29] Peer audio RTP is at port 192.168.1.85:30016 [Mar 10 12:07:29] Found description format PCMU for ID 0 [Mar 10 12:07:29] Found description format telephone-event for ID 101 [Mar 10 12:07:29] Got unsupported a:fmtp in SDP offer [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:5123 process_sdp: T38 state changed to 0 on channel SIP/523-087dd9a8 [Mar 10 12:07:29] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 10 12:07:29] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 10 12:07:29] Peer audio RTP is at port 192.168.1.85:30016 [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:5203 process_sdp: We're settling with these formats: 0x4 (ulaw) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:5210 process_sdp: We have an owner, now see if we need to change this call [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:3001 update_call_counter: Updating call counter for outgoing call [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:7903 build_route: build_route: Retaining previous route: [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:11752 handle_response_invite: Strange... The other side of the bridge does not have a udptl struct [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:11757 handle_response_invite: T38 state changed to 0 on channel SIP [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:11760 handle_response_invite: T38 state changed to 0 on channel SIP/523-087dd9a8 [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:5637 reqprep: Strict routing enforced for session 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 [Mar 10 12:07:29] set_destination: Parsing for address/port to send to [Mar 10 12:07:29] set_destination: set destination to 192.168.1.85, port 5060 [Mar 10 12:07:29] Transmitting (no NAT) to 192.168.1.85:5060: ACK sip:523@192.168.1.85:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK32e7c40d From: "593 593" ;tag=as628c2367 To: ;tag=a0b2f919869f2cd75f7879cc9d8200ec Contact: Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 10 12:07:29] <--- SIP read from 192.168.1.83:5060 ---> SIP/2.0 200 OK Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 CSeq: 102 INVITE From: 523 ;tag=as43feeb80 To: 593 ;tag=48204f4e1c15803 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK150e6c3d Content-Length: 266 Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE Content-Type: application/sdp Supported: replaces Contact: 593 User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 v=0 o=MxSIP 0 1143068287 IN IP4 192.168.1.83 s=SIP Call c=IN IP4 192.168.1.83 t=0 0 m=audio 51000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:on - - - - <-------------> [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: SIP/2.0 200 OK (14) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 (54) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 102 INVITE (16) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: From: 523 ;tag=as43feeb80 (53) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: To: 593 ;tag=48204f4e1c15803 (56) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK150e6c3d (58) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Content-Length: 266 (19) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE (55) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Content-Type: application/sdp (29) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Supported: replaces (19) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Contact: 593 (40) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 (42) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: (0) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: o=MxSIP 0 1143068287 IN IP4 192.168.1.83 (40) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: s=SIP Call (10) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: c=IN IP4 192.168.1.83 (21) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: m=audio 51000 RTP/AVP 0 8 101 (29) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-15 (15) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=sendrecv (10) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=silenceSupp:on - - - - (24) [Mar 10 12:07:29] --- (12 headers 13 lines) --- [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:2069 __sip_ack: Acked pending invite 102 [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #178 [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on 'a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83' of Request 102: Match Not Found [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:11619 handle_response_invite: SIP response 200 to RE-invite on outgoing call a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 [Mar 10 12:07:29] Found RTP audio format 0 [Mar 10 12:07:29] Found RTP audio format 8 [Mar 10 12:07:29] Found RTP audio format 101 [Mar 10 12:07:29] Peer audio RTP is at port 192.168.1.83:51000 [Mar 10 12:07:29] Found description format PCMU for ID 0 [Mar 10 12:07:29] Found description format PCMA for ID 8 [Mar 10 12:07:29] Found description format telephone-event for ID 101 [Mar 10 12:07:29] Got unsupported a:fmtp in SDP offer [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:5123 process_sdp: T38 state changed to 0 on channel SIP/593-b77052f8 [Mar 10 12:07:29] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) [Mar 10 12:07:29] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 10 12:07:29] Peer audio RTP is at port 192.168.1.83:51000 [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:5203 process_sdp: We're settling with these formats: 0xc (ulaw|alaw) [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:5210 process_sdp: We have an owner, now see if we need to change this call [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:3001 update_call_counter: Updating call counter for incoming call [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:7964 build_route: build_route: Contact hop: 593 [Mar 10 12:07:29] list_route: hop: [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:11752 handle_response_invite: Strange... The other side of the bridge does not have a udptl struct [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:11757 handle_response_invite: T38 state changed to 0 on channel SIP [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:11760 handle_response_invite: T38 state changed to 0 on channel SIP/593-b77052f8 [Mar 10 12:07:29] DEBUG[18555]: chan_sip.c:5637 reqprep: Strict routing enforced for session a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 [Mar 10 12:07:29] set_destination: Parsing for address/port to send to [Mar 10 12:07:29] set_destination: set destination to 192.168.1.83, port 5060 [Mar 10 12:07:29] Transmitting (no NAT) to 192.168.1.83:5060: ACK sip:593@192.168.1.83:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK51137b1f From: 523 ;tag=as43feeb80 To: 593 ;tag=48204f4e1c15803 Contact: Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 10 12:07:31] <--- SIP read from 192.168.1.86:5060 ---> <-------------> [Mar 10 12:07:31] --- (0 headers 0 lines) Nat keepalive --- [Mar 10 12:07:33] <--- SIP read from 192.168.1.61:5060 ---> REGISTER sip:24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bKc6e91d87b23962970fabb34f7bb9d8972 Call-ID: 55ae6647678c@24.123.23.170 CSeq: 61345 REGISTER From: ;tag=3556b65341152693ba6be3bbe481fdc3 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="46c2e0ca", algorithm=MD5, uri="sip:24.123.23.170", username="524", response="a22007204ef5ee692440c2eeefa9eab8" <-------------> [Mar 10 12:07:33] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:07:33] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bKc6e91d87b23962970fabb34f7bb9d8972 (82) [Mar 10 12:07:33] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Call-ID: 55ae6647678c@24.123.23.170 (35) [Mar 10 12:07:33] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: CSeq: 61345 REGISTER (20) [Mar 10 12:07:33] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=3556b65341152693ba6be3bbe481fdc3 (66) [Mar 10 12:07:33] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:07:33] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:07:33] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:07:33] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:07:33] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:07:33] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:07:33] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:07:33] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="46c2e0ca", algorithm=MD5, uri="sip:24.123.23.170", username="524", response="a22007204ef5ee692440c2eeefa9eab8" (167) [Mar 10 12:07:33] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:07:33] --- (13 headers 0 lines) --- [Mar 10 12:07:33] DEBUG[18555]: chan_sip.c:4308 sip_alloc: Allocating new SIP dialog for 55ae6647678c@24.123.23.170 - REGISTER (No RTP) [Mar 10 12:07:33] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:07:33] Using latest REGISTER request as basis request [Mar 10 12:07:33] Sending to 192.168.1.61 : 5060 (no NAT) [Mar 10 12:07:33] <--- Transmitting (no NAT) to 192.168.1.61:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bKc6e91d87b23962970fabb34f7bb9d8972;received=192.168.1.61 From: ;tag=3556b65341152693ba6be3bbe481fdc3 To: Call-ID: 55ae6647678c@24.123.23.170 CSeq: 61345 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:33] <--- Transmitting (no NAT) to 192.168.1.61:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bKc6e91d87b23962970fabb34f7bb9d8972;received=192.168.1.61 From: ;tag=3556b65341152693ba6be3bbe481fdc3 To: ;tag=as28d4ded5 Call-ID: 55ae6647678c@24.123.23.170 CSeq: 61345 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7ed20082" Content-Length: 0 <------------> [Mar 10 12:07:33] Scheduling destruction of SIP dialog '55ae6647678c@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:07:33] <--- SIP read from 192.168.1.61:5060 ---> REGISTER sip:24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bKj11e35fdf01063be78d21a82f3bcfbae8 CSeq: 61346 REGISTER Call-ID: 55ae6647678c@24.123.23.170 From: ;tag=3556b65341152693ba6be3bbe481fdc3 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="7ed20082", algorithm=MD5, uri="sip:24.123.23.170", username="524", response="9eaccb2dc169d282647553a96f9a1d22" <-------------> [Mar 10 12:07:33] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:07:33] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bKj11e35fdf01063be78d21a82f3bcfbae8 (82) [Mar 10 12:07:33] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 61346 REGISTER (20) [Mar 10 12:07:33] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Call-ID: 55ae6647678c@24.123.23.170 (35) [Mar 10 12:07:33] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=3556b65341152693ba6be3bbe481fdc3 (66) [Mar 10 12:07:33] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:07:33] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:07:33] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:07:33] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:07:33] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:07:33] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:07:33] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:07:33] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="7ed20082", algorithm=MD5, uri="sip:24.123.23.170", username="524", response="9eaccb2dc169d282647553a96f9a1d22" (167) [Mar 10 12:07:33] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:07:33] --- (13 headers 0 lines) --- [Mar 10 12:07:33] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:07:33] Using latest REGISTER request as basis request [Mar 10 12:07:33] Sending to 192.168.1.61 : 5060 (no NAT) [Mar 10 12:07:33] <--- Transmitting (no NAT) to 192.168.1.61:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bKj11e35fdf01063be78d21a82f3bcfbae8;received=192.168.1.61 From: ;tag=3556b65341152693ba6be3bbe481fdc3 To: Call-ID: 55ae6647678c@24.123.23.170 CSeq: 61346 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:33] <--- Transmitting (no NAT) to 192.168.1.61:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bKj11e35fdf01063be78d21a82f3bcfbae8;received=192.168.1.61 From: ;tag=3556b65341152693ba6be3bbe481fdc3 To: ;tag=as28d4ded5 Call-ID: 55ae6647678c@24.123.23.170 CSeq: 61346 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Sat, 10 Mar 2007 17:07:33 GMT Content-Length: 0 <------------> [Mar 10 12:07:33] DEBUG[18555]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/524 [Mar 10 12:07:33] Scheduling destruction of SIP dialog '55ae6647678c@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:07:33] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 524 [Mar 10 12:07:33] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 524 [Mar 10 12:07:33] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/524 - state 1 (Not in use) [Mar 10 12:07:33] DEBUG[18597]: app_queue.c:546 changethread: Device 'SIP/524' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Mar 10 12:07:35] <--- SIP read from 192.168.1.95:5060 ---> REGISTER sip:24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.95:5060;branch=z9hG4bKsa45f13089c79afb70353a435e3f2bc91 Call-ID: 55ae66469f11@24.123.23.170 CSeq: 49844 REGISTER From: ;tag=5b365ed2a85cbf72f8086096c2e8e2da 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="15617ed6", algorithm=MD5, uri="sip:24.123.23.170", username="522", response="8fbd51d011eb45a6226943ddb434a39d" <-------------> [Mar 10 12:07:35] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:07:35] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.95:5060;branch=z9hG4bKsa45f13089c79afb70353a435e3f2bc91 (82) [Mar 10 12:07:35] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Call-ID: 55ae66469f11@24.123.23.170 (35) [Mar 10 12:07:35] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: CSeq: 49844 REGISTER (20) [Mar 10 12:07:35] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=5b365ed2a85cbf72f8086096c2e8e2da (66) [Mar 10 12:07:35] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:07:35] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:07:35] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:07:35] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:07:35] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:07:35] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:07:35] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:07:35] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="15617ed6", algorithm=MD5, uri="sip:24.123.23.170", username="522", response="8fbd51d011eb45a6226943ddb434a39d" (167) [Mar 10 12:07:35] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:07:35] --- (13 headers 0 lines) --- [Mar 10 12:07:35] DEBUG[18555]: chan_sip.c:4308 sip_alloc: Allocating new SIP dialog for 55ae66469f11@24.123.23.170 - REGISTER (No RTP) [Mar 10 12:07:35] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:07:35] Using latest REGISTER request as basis request [Mar 10 12:07:35] Sending to 192.168.1.95 : 5060 (no NAT) [Mar 10 12:07:35] <--- Transmitting (no NAT) to 192.168.1.95:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.95:5060;branch=z9hG4bKsa45f13089c79afb70353a435e3f2bc91;received=192.168.1.95 From: ;tag=5b365ed2a85cbf72f8086096c2e8e2da To: Call-ID: 55ae66469f11@24.123.23.170 CSeq: 49844 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:35] <--- Transmitting (no NAT) to 192.168.1.95:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.95:5060;branch=z9hG4bKsa45f13089c79afb70353a435e3f2bc91;received=192.168.1.95 From: ;tag=5b365ed2a85cbf72f8086096c2e8e2da To: ;tag=as7df448b1 Call-ID: 55ae66469f11@24.123.23.170 CSeq: 49844 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6edabc0f" Content-Length: 0 <------------> [Mar 10 12:07:35] Scheduling destruction of SIP dialog '55ae66469f11@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:07:35] <--- SIP read from 192.168.1.95:5060 ---> REGISTER sip:24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.95:5060;branch=z9hG4bKf499adbaf77746f27a1ca964646b45236 CSeq: 49845 REGISTER Call-ID: 55ae66469f11@24.123.23.170 From: ;tag=5b365ed2a85cbf72f8086096c2e8e2da 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="6edabc0f", algorithm=MD5, uri="sip:24.123.23.170", username="522", response="e94354fd6ac993867af453b3fc8e729e" <-------------> [Mar 10 12:07:35] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:07:35] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.95:5060;branch=z9hG4bKf499adbaf77746f27a1ca964646b45236 (82) [Mar 10 12:07:35] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 49845 REGISTER (20) [Mar 10 12:07:35] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Call-ID: 55ae66469f11@24.123.23.170 (35) [Mar 10 12:07:35] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=5b365ed2a85cbf72f8086096c2e8e2da (66) [Mar 10 12:07:35] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:07:35] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:07:35] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:07:35] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:07:35] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:07:35] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:07:35] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:07:35] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="6edabc0f", algorithm=MD5, uri="sip:24.123.23.170", username="522", response="e94354fd6ac993867af453b3fc8e729e" (167) [Mar 10 12:07:35] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:07:35] --- (13 headers 0 lines) --- [Mar 10 12:07:35] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:07:35] Using latest REGISTER request as basis request [Mar 10 12:07:35] Sending to 192.168.1.95 : 5060 (no NAT) [Mar 10 12:07:35] <--- Transmitting (no NAT) to 192.168.1.95:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.95:5060;branch=z9hG4bKf499adbaf77746f27a1ca964646b45236;received=192.168.1.95 From: ;tag=5b365ed2a85cbf72f8086096c2e8e2da To: Call-ID: 55ae66469f11@24.123.23.170 CSeq: 49845 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:35] <--- Transmitting (no NAT) to 192.168.1.95:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.95:5060;branch=z9hG4bKf499adbaf77746f27a1ca964646b45236;received=192.168.1.95 From: ;tag=5b365ed2a85cbf72f8086096c2e8e2da To: ;tag=as7df448b1 Call-ID: 55ae66469f11@24.123.23.170 CSeq: 49845 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Sat, 10 Mar 2007 17:07:35 GMT Content-Length: 0 <------------> [Mar 10 12:07:35] DEBUG[18555]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/522 [Mar 10 12:07:35] Scheduling destruction of SIP dialog '55ae66469f11@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:07:35] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 522 [Mar 10 12:07:35] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 522 [Mar 10 12:07:35] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/522 - state 1 (Not in use) [Mar 10 12:07:35] DEBUG[18598]: 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. [Mar 10 12:07:36] <--- SIP read from 192.168.1.97:5060 ---> INVITE sip:523@24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.97:5060;rport;branch=z9hG4bK3800469616 From: "Display Name" ;tag=2534858470 To: Call-ID: 2919426183@192.168.1.97 CSeq: 1000 INVITE Contact: max-forwards: 70 user-agent: UTSTARCOM F1000/Device ID-0007ba26174b Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, MESSAGE, REGISTER, REFER, PING, NOTIFY, INFO Content-Type: application/sdp Content-Length: 298 v=0 o=551 123456 654332 IN IP4 192.168.1.97 s=none c=IN IP4 192.168.1.97 t=0 0 m=audio 10010 RTP/AVP 0 8 18 2 101 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729A/8000 a=fmtp:18 annexb=no a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: INVITE sip:523@24.123.23.170 SIP/2.0 (36) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.97:5060;rport;branch=z9hG4bK3800469616 (65) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: From: "Display Name" ;tag=2534858470 (59) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: (27) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 2919426183@192.168.1.97 (32) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: CSeq: 1000 INVITE (17) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: (36) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: max-forwards: 70 (16) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: user-agent: UTSTARCOM F1000/Device ID-0007ba26174b (50) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, MESSAGE, REGISTER, REFER, PING, NOTIFY, INFO (86) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Content-Type: application/sdp (29) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Content-Length: 298 (21) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: (0) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: o=551 123456 654332 IN IP4 192.168.1.97 (39) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: s=none (6) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: c=IN IP4 192.168.1.97 (21) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: m=audio 10010 RTP/AVP 0 8 18 2 101 (34) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:18 G729A/8000 (22) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=fmtp:18 annexb=no (19) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:2 G726-32/8000 (23) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-15 (15) [Mar 10 12:07:36] --- (12 headers 14 lines) --- [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:2573 do_setnat: Setting NAT on RTP to Off [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:2578 do_setnat: Setting NAT on VRTP to Off [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4308 sip_alloc: Allocating new SIP dialog for 2919426183@192.168.1.97 - INVITE (With RTP) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Mar 10 12:07:36] Sending to 192.168.1.97 : 5060 (NAT) [Mar 10 12:07:36] Using INVITE request as basis request - 2919426183@192.168.1.97 [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:2573 do_setnat: Setting NAT on RTP to On [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:2578 do_setnat: Setting NAT on VRTP to On [Mar 10 12:07:36] <--- Reliably Transmitting (NAT) to 192.168.1.97:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.97:5060;branch=z9hG4bK3800469616;received=192.168.1.97;rport=5060 From: "Display Name" ;tag=2534858470 To: ;tag=as6debc37d Call-ID: 2919426183@192.168.1.97 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="77da93b9" Content-Length: 0 <------------> [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #185 [Mar 10 12:07:36] Scheduling destruction of SIP dialog '2919426183@192.168.1.97' in 32000 ms (Method: INVITE) [Mar 10 12:07:36] Found user '551' [Mar 10 12:07:36] <--- SIP read from 192.168.1.97:5060 ---> <-------------> [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: (0) [Mar 10 12:07:36] --- (0 headers 0 lines) Nat keepalive --- [Mar 10 12:07:36] <--- SIP read from 192.168.1.97:5060 ---> ACK sip:523@24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.97:5060;rport;branch=z9hG4bK3800469616 From: "Display Name" ;tag=2534858470 To: ;tag=as6debc37d Call-ID: 2919426183@192.168.1.97 CSeq: 1000 ACK Content-Length: 0 <-------------> [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: ACK sip:523@24.123.23.170 SIP/2.0 (33) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.97:5060;rport;branch=z9hG4bK3800469616 (65) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: From: "Display Name" ;tag=2534858470 (59) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: ;tag=as6debc37d (42) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 2919426183@192.168.1.97 (32) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: CSeq: 1000 ACK (14) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Content-Length: 0 (17) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: (0) [Mar 10 12:07:36] --- (7 headers 0 lines) --- [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received ACK (6) - Command in SIP ACK [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #185 [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '2919426183@192.168.1.97' of Response 1000: Match Not Found [Mar 10 12:07:36] <--- SIP read from 192.168.1.97:5060 ---> INVITE sip:523@24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.97:5060;rport;branch=z9hG4bK1788404449 From: "Display Name" ;tag=2534858470 To: Call-ID: 2919426183@192.168.1.97 CSeq: 1001 INVITE Contact: Proxy-Authorization: Digest username="551", realm="asterisk", nonce="77da93b9", uri="sip:523@24.123.23.170", response="c1e843ca52088e3785f62c66c9fd09a5", algorithm=MD5 max-forwards: 70 user-agent: UTSTARCOM F1000/Device ID-0007ba26174b Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, MESSAGE, REGISTER, REFER, PING, NOTIFY, INFO Content-Type: application/sdp Content-Length: 298 v=0 o=551 123456 654332 IN IP4 192.168.1.97 s=none c=IN IP4 192.168.1.97 t=0 0 m=audio 10010 RTP/AVP 0 8 18 2 101 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729A/8000 a=fmtp:18 annexb=no a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: INVITE sip:523@24.123.23.170 SIP/2.0 (36) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.97:5060;rport;branch=z9hG4bK1788404449 (65) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: From: "Display Name" ;tag=2534858470 (59) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: (27) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 2919426183@192.168.1.97 (32) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: CSeq: 1001 INVITE (17) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: (36) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Proxy-Authorization: Digest username="551", realm="asterisk", nonce="77da93b9", uri="sip:523@24.123.23.170", response="c1e843ca52088e3785f62c66c9fd09a5", algorithm=MD5 (167) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: max-forwards: 70 (16) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: user-agent: UTSTARCOM F1000/Device ID-0007ba26174b (50) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, MESSAGE, REGISTER, REFER, PING, NOTIFY, INFO (86) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Content-Type: application/sdp (29) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Content-Length: 298 (21) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: o=551 123456 654332 IN IP4 192.168.1.97 (39) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: s=none (6) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: c=IN IP4 192.168.1.97 (21) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: m=audio 10010 RTP/AVP 0 8 18 2 101 (34) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:18 G729A/8000 (22) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=fmtp:18 annexb=no (19) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:2 G726-32/8000 (23) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-15 (15) [Mar 10 12:07:36] --- (13 headers 14 lines) --- [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Mar 10 12:07:36] Sending to 192.168.1.97 : 5060 (NAT) [Mar 10 12:07:36] Using INVITE request as basis request - 2919426183@192.168.1.97 [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:2573 do_setnat: Setting NAT on RTP to On [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:2578 do_setnat: Setting NAT on VRTP to On [Mar 10 12:07:36] Found user '551' [Mar 10 12:07:36] Found RTP audio format 0 [Mar 10 12:07:36] Found RTP audio format 8 [Mar 10 12:07:36] Found RTP audio format 18 [Mar 10 12:07:36] Found RTP audio format 2 [Mar 10 12:07:36] Found RTP audio format 101 [Mar 10 12:07:36] Peer audio RTP is at port 192.168.1.97:10010 [Mar 10 12:07:36] Found description format PCMU for ID 0 [Mar 10 12:07:36] Found description format PCMA for ID 8 [Mar 10 12:07:36] Found description format G729A for ID 18 [Mar 10 12:07:36] Got unsupported a:fmtp in SDP offer [Mar 10 12:07:36] Found description format G726-32 for ID 2 [Mar 10 12:07:36] Found description format telephone-event for ID 101 [Mar 10 12:07:36] Got unsupported a:fmtp in SDP offer [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:5123 process_sdp: T38 state changed to 0 on channel [Mar 10 12:07:36] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x90c (ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) [Mar 10 12:07:36] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 10 12:07:36] Peer audio RTP is at port 192.168.1.97:10010 [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:5203 process_sdp: We're settling with these formats: 0xc (ulaw|alaw) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:13378 handle_request_invite: Checking SIP call limits for device 551 [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:3001 update_call_counter: Updating call counter for incoming call [Mar 10 12:07:36] Looking for 523 in smvoice-sip (domain 24.123.23.170) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:3803 sip_new: *** Our native formats are 0x4 (ulaw) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:3804 sip_new: *** Joint capabilities are 0xc (ulaw|alaw) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:3805 sip_new: *** Our capabilities are 0xe (gsm|ulaw|alaw) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:3806 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:3829 sip_new: This channel will not be able to handle video. [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:7964 build_route: build_route: Contact hop: [Mar 10 12:07:36] list_route: hop: [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:13453 handle_request_invite: SIP/551-b770d980: New call is still down.... Trying... [Mar 10 12:07:36] <--- Transmitting (NAT) to 192.168.1.97:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.97:5060;branch=z9hG4bK1788404449;received=192.168.1.97;rport=5060 From: "Display Name" ;tag=2534858470 To: Call-ID: 2919426183@192.168.1.97 CSeq: 1001 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:36] DEBUG[18555]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/551-b770d980 [Mar 10 12:07:36] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 551 [Mar 10 12:07:36] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 551 [Mar 10 12:07:36] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/551 - state 1 (Not in use) [Mar 10 12:07:36] DEBUG[18599]: pbx.c:1791 pbx_extension_helper: Launching 'NoOp' [Mar 10 12:07:36] -- Executing [523@smvoice-sip:1] NoOp("SIP/551-b770d980", "5xx") in new stack [Mar 10 12:07:36] DEBUG[18599]: pbx.c:1791 pbx_extension_helper: Launching 'Set' [Mar 10 12:07:36] -- Executing [523@smvoice-sip:2] Set("SIP/551-b770d980", "SMVOICE_CONTEXT_EXTEN=523") in new stack [Mar 10 12:07:36] DEBUG[18599]: pbx.c:1791 pbx_extension_helper: Launching 'AGI' [Mar 10 12:07:36] -- Executing [523@smvoice-sip:3] AGI("SIP/551-b770d980", "smvoice|-company|demonstration|-digium_asterisk|-asterisk_callat_forwarding|523") in new stack [Mar 10 12:07:36] -- Launched AGI Script /var/lib/asterisk/agi-bin/smvoice [Mar 10 12:07:36] DEBUG[18600]: app_queue.c:546 changethread: Device 'SIP/551' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Mar 10 12:07:36] -- AGI Script smvoice completed, returning 0 [Mar 10 12:07:36] DEBUG[18599]: pbx.c:1712 pbx_substitute_variables_helper_full: Expression result is '0' [Mar 10 12:07:36] DEBUG[18599]: pbx.c:1791 pbx_extension_helper: Launching 'GotoIf' [Mar 10 12:07:36] -- Executing [523@smvoice-sip:4] GotoIf("SIP/551-b770d980", "0?INVALID|1") in new stack [Mar 10 12:07:36] DEBUG[18599]: pbx.c:5969 pbx_builtin_gotoif: Not taking any branch [Mar 10 12:07:36] DEBUG[18599]: pbx.c:1712 pbx_substitute_variables_helper_full: Expression result is '0' [Mar 10 12:07:36] DEBUG[18599]: pbx.c:1791 pbx_extension_helper: Launching 'GotoIf' [Mar 10 12:07:36] -- Executing [523@smvoice-sip:5] GotoIf("SIP/551-b770d980", "0?_5XX-NOANSWER|1") in new stack [Mar 10 12:07:36] DEBUG[18599]: pbx.c:5969 pbx_builtin_gotoif: Not taking any branch [Mar 10 12:07:36] DEBUG[18599]: pbx.c:1791 pbx_extension_helper: Launching 'Dial' [Mar 10 12:07:36] -- Executing [523@smvoice-sip:6] Dial("SIP/551-b770d980", "SIP/523|20|") in new stack [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:15267 sip_request_call: Asked to create a SIP channel with formats: 0x4 (ulaw) [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:4308 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:2573 do_setnat: Setting NAT on RTP to Off [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:3803 sip_new: *** Our native formats are 0x4 (ulaw) [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:3804 sip_new: *** Joint capabilities are 0x0 (nothing) [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:3805 sip_new: *** Our capabilities are 0xe (gsm|ulaw|alaw) [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:3806 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:3808 sip_new: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:3829 sip_new: This channel will not be able to handle video. [Mar 10 12:07:36] DEBUG[18599]: rtp.c:1579 ast_rtp_make_compatible: Seeded SDP of 'SIP/523-087e2e08' with that of 'SIP/551-b770d980' [Mar 10 12:07:36] DEBUG[18599]: channel.c:3301 ast_channel_inherit_variables: Not copying variable STACK-smvoice-sip-523-6. [Mar 10 12:07:36] DEBUG[18599]: channel.c:3301 ast_channel_inherit_variables: Not copying variable STACK-smvoice-sip-523-5. [Mar 10 12:07:36] DEBUG[18599]: channel.c:3301 ast_channel_inherit_variables: Not copying variable STACK-smvoice-sip-523-4. [Mar 10 12:07:36] DEBUG[18599]: channel.c:3301 ast_channel_inherit_variables: Not copying variable AGISTATUS. [Mar 10 12:07:36] DEBUG[18599]: channel.c:3301 ast_channel_inherit_variables: Not copying variable SMVOICE_WORK_EXTENSION. [Mar 10 12:07:36] DEBUG[18599]: channel.c:3301 ast_channel_inherit_variables: Not copying variable SMVOICE_EXTENSION. [Mar 10 12:07:36] DEBUG[18599]: channel.c:3301 ast_channel_inherit_variables: Not copying variable SMVOICE_PIN. [Mar 10 12:07:36] DEBUG[18599]: channel.c:3301 ast_channel_inherit_variables: Not copying variable SMVOICE_CALLAT. [Mar 10 12:07:36] DEBUG[18599]: channel.c:3301 ast_channel_inherit_variables: Not copying variable STACK-smvoice-sip-523-3. [Mar 10 12:07:36] DEBUG[18599]: channel.c:3301 ast_channel_inherit_variables: Not copying variable SMVOICE_CONTEXT_EXTEN. [Mar 10 12:07:36] DEBUG[18599]: channel.c:3301 ast_channel_inherit_variables: Not copying variable STACK-smvoice-sip-523-2. [Mar 10 12:07:36] DEBUG[18599]: channel.c:3301 ast_channel_inherit_variables: Not copying variable STACK-smvoice-sip-523-1. [Mar 10 12:07:36] DEBUG[18599]: channel.c:3301 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Mar 10 12:07:36] DEBUG[18599]: channel.c:3301 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT. [Mar 10 12:07:36] DEBUG[18599]: channel.c:3301 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Mar 10 12:07:36] DEBUG[18599]: channel.c:3301 ast_channel_inherit_variables: Not copying variable SIPURI. [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:2828 sip_call: Outgoing Call for 523 [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:3001 update_call_counter: Updating call counter for outgoing call [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:2843 sip_call: Our T38 capability (0), joint T38 capability (0) [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:6182 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: False [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:6183 add_sdp: ** Our prefcodec: 0x4 (ulaw) [Mar 10 12:07:36] Audio is at 24.123.23.170 port 11786 [Mar 10 12:07:36] Adding codec 0x4 (ulaw) to SDP [Mar 10 12:07:36] Adding codec 0x8 (alaw) to SDP [Mar 10 12:07:36] Adding codec 0x2 (gsm) to SDP [Mar 10 12:07:36] Adding non-codec 0x1 (telephone-event) to SDP [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:6314 add_sdp: -- Done with adding codecs to SDP [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:6359 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 0: INVITE sip:523@192.168.1.85:5060 SIP/2.0 (40) [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK14f10e49 (58) [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 2: From: "551 551" ;tag=as59744546 (54) [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 3: To: (31) [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 4: Contact: (32) [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 5: Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 (55) [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 6: CSeq: 102 INVITE (16) [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 9: Date: Sat, 10 Mar 2007 17:07:36 GMT (35) [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 11: Supported: replaces (19) [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 12: Content-Type: application/sdp (29) [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 13: Content-Length: 289 (19) [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 14: (0) [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:4603 parse_request: Line: o=root 18522 18522 IN IP4 24.123.23.170 (39) [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:4603 parse_request: Line: s=session (9) [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:4603 parse_request: Line: c=IN IP4 24.123.23.170 (22) [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:4603 parse_request: Line: m=audio 11786 RTP/AVP 0 8 3 101 (31) [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:4603 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:4603 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-16 (15) [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:4603 parse_request: Line: a=silenceSupp:off - - - - (25) [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:4603 parse_request: Line: a=sendrecv (10) [Mar 10 12:07:36] Reliably Transmitting (no NAT) to 192.168.1.85:5060: INVITE sip:523@192.168.1.85:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK14f10e49 From: "551 551" ;tag=as59744546 To: Contact: Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sat, 10 Mar 2007 17:07:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 289 v=0 o=root 18522 18522 IN IP4 24.123.23.170 s=session c=IN IP4 24.123.23.170 t=0 0 m=audio 11786 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 --- [Mar 10 12:07:36] DEBUG[18599]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #187 [Mar 10 12:07:36] -- Called 523 [Mar 10 12:07:36] <--- SIP read from 192.168.1.85:5060 ---> SIP/2.0 100 Trying To: ;tag=fb7c7b32eac4985d4d8676d2196e3753 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK14f10e49 From: "551 551" ;tag=as59744546 Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 CSeq: 102 INVITE Content-Length: 0 Supported: sip-cc, sip-cc-01, replaces, timer Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK <-------------> [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: SIP/2.0 100 Trying (18) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: To: ;tag=fb7c7b32eac4985d4d8676d2196e3753 (68) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK14f10e49 (58) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: From: "551 551" ;tag=as59744546 (54) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 (55) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: CSeq: 102 INVITE (16) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Content-Length: 0 (17) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: (0) [Mar 10 12:07:36] --- (9 headers 0 lines) --- [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:2120 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #187 - INVITE (got response) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:2129 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170' Request 102: Found [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:11621 handle_response_invite: SIP response 100 to standard invite [Mar 10 12:07:36] <--- SIP read from 192.168.1.66:5060 ---> REGISTER sip:24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bKk6244627cc204cffe600bb5b4fd1fe7f5 Call-ID: 55ae6647666e@24.123.23.170 CSeq: 24036 REGISTER From: ;tag=693d936234208f506e2b3880e0beb9f4 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="5caef355", algorithm=MD5, uri="sip:24.123.23.170", username="526", response="f9b047fc785d64e185405f14f56e2b48" <-------------> [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bKk6244627cc204cffe600bb5b4fd1fe7f5 (82) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Call-ID: 55ae6647666e@24.123.23.170 (35) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: CSeq: 24036 REGISTER (20) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=693d936234208f506e2b3880e0beb9f4 (66) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="5caef355", algorithm=MD5, uri="sip:24.123.23.170", username="526", response="f9b047fc785d64e185405f14f56e2b48" (167) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:07:36] --- (13 headers 0 lines) --- [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4308 sip_alloc: Allocating new SIP dialog for 55ae6647666e@24.123.23.170 - REGISTER (No RTP) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:07:36] Using latest REGISTER request as basis request [Mar 10 12:07:36] Sending to 192.168.1.66 : 5060 (no NAT) [Mar 10 12:07:36] <--- Transmitting (no NAT) to 192.168.1.66:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bKk6244627cc204cffe600bb5b4fd1fe7f5;received=192.168.1.66 From: ;tag=693d936234208f506e2b3880e0beb9f4 To: Call-ID: 55ae6647666e@24.123.23.170 CSeq: 24036 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:36] <--- Transmitting (no NAT) to 192.168.1.66:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bKk6244627cc204cffe600bb5b4fd1fe7f5;received=192.168.1.66 From: ;tag=693d936234208f506e2b3880e0beb9f4 To: ;tag=as1b061ad1 Call-ID: 55ae6647666e@24.123.23.170 CSeq: 24036 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3dd40c53" Content-Length: 0 <------------> [Mar 10 12:07:36] Scheduling destruction of SIP dialog '55ae6647666e@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:07:36] <--- SIP read from 192.168.1.85:5060 ---> SIP/2.0 180 Ringing To: ;tag=fb7c7b32eac4985d4d8676d2196e3753 Contact: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK14f10e49 From: "551 551" ;tag=as59744546 Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 CSeq: 102 INVITE Content-Length: 0 Supported: sip-cc, sip-cc-01, replaces, timer Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK <-------------> [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: SIP/2.0 180 Ringing (19) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: To: ;tag=fb7c7b32eac4985d4d8676d2196e3753 (68) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Contact: (36) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK14f10e49 (58) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: "551 551" ;tag=as59744546 (54) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 (55) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: CSeq: 102 INVITE (16) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Content-Length: 0 (17) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: (0) [Mar 10 12:07:36] --- (10 headers 0 lines) --- [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:2129 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170' Request 102: Found [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:11621 handle_response_invite: SIP response 180 to standard invite [Mar 10 12:07:36] DEBUG[18555]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/523-087e2e08 [Mar 10 12:07:36] -- SIP/523-087e2e08 is ringing [Mar 10 12:07:36] DEBUG[18599]: rtp.c:1514 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/551-b770d980' with that of 'SIP/523-087e2e08' [Mar 10 12:07:36] <--- Transmitting (NAT) to 192.168.1.97:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.97:5060;branch=z9hG4bK1788404449;received=192.168.1.97;rport=5060 From: "Display Name" ;tag=2534858470 To: ;tag=as08598f69 Call-ID: 2919426183@192.168.1.97 CSeq: 1001 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:36] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 523 [Mar 10 12:07:36] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 523 [Mar 10 12:07:36] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/523 - state 1 (Not in use) [Mar 10 12:07:36] DEBUG[18602]: 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. [Mar 10 12:07:36] <--- SIP read from 192.168.1.66:5060 ---> REGISTER sip:24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bKn472281de861ea0924355dad7017e16d5 CSeq: 24037 REGISTER Call-ID: 55ae6647666e@24.123.23.170 From: ;tag=693d936234208f506e2b3880e0beb9f4 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="3dd40c53", algorithm=MD5, uri="sip:24.123.23.170", username="526", response="37a08fbc5635f2b1ec490a7d3e97a29e" <-------------> [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bKn472281de861ea0924355dad7017e16d5 (82) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 24037 REGISTER (20) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Call-ID: 55ae6647666e@24.123.23.170 (35) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=693d936234208f506e2b3880e0beb9f4 (66) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="3dd40c53", algorithm=MD5, uri="sip:24.123.23.170", username="526", response="37a08fbc5635f2b1ec490a7d3e97a29e" (167) [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:07:36] --- (13 headers 0 lines) --- [Mar 10 12:07:36] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:07:36] Using latest REGISTER request as basis request [Mar 10 12:07:36] Sending to 192.168.1.66 : 5060 (no NAT) [Mar 10 12:07:36] <--- Transmitting (no NAT) to 192.168.1.66:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bKn472281de861ea0924355dad7017e16d5;received=192.168.1.66 From: ;tag=693d936234208f506e2b3880e0beb9f4 To: Call-ID: 55ae6647666e@24.123.23.170 CSeq: 24037 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:36] <--- Transmitting (no NAT) to 192.168.1.66:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bKn472281de861ea0924355dad7017e16d5;received=192.168.1.66 From: ;tag=693d936234208f506e2b3880e0beb9f4 To: ;tag=as1b061ad1 Call-ID: 55ae6647666e@24.123.23.170 CSeq: 24037 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Sat, 10 Mar 2007 17:07:36 GMT Content-Length: 0 <------------> [Mar 10 12:07:36] DEBUG[18555]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/526 [Mar 10 12:07:36] Scheduling destruction of SIP dialog '55ae6647666e@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:07:36] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 526 [Mar 10 12:07:36] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 526 [Mar 10 12:07:36] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/526 - state 1 (Not in use) [Mar 10 12:07:36] DEBUG[18603]: 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. [Mar 10 12:07:37] <--- SIP read from 192.168.1.85:5060 ---> INVITE sip:593@24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKc128ad6bf7a12c24938883bffed31855b Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 388020 INVITE From: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec To: ;tag=as628c2367 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=- 18522 204406 IN IP4 192.168.1.85 s=session c=IN IP4 0.0.0.0 t=0 0 m=audio 30016 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 <-------------> [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: INVITE sip:593@24.123.23.170 SIP/2.0 (36) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKc128ad6bf7a12c24938883bffed31855b (82) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 (55) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: CSeq: 388020 INVITE (19) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec (74) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: ;tag=as628c2367 (42) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: (36) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Content-Type: application/sdp (29) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Max-Forwards: 70 (16) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Content-Length: 242 (19) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: o=- 18522 204406 IN IP4 192.168.1.85 (36) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: s=session (9) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: c=IN IP4 0.0.0.0 (16) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: m=audio 30016 RTP/AVP 0 101 (27) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-15,16,17,18 (24) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=silenceSupp:off - - - - (25) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=sendonly (10) [Mar 10 12:07:37] --- (13 headers 12 lines) --- [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:1678 parse_sip_options: Begin: parsing SIP "Supported: sip-cc, sip-cc-01, replaces, timer" [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:1686 parse_sip_options: Found SIP option: -sip-cc- [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:1700 parse_sip_options: Found no match for SIP option: sip-cc (Please file bug report!) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:1686 parse_sip_options: Found SIP option: -sip-cc-01- [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:1700 parse_sip_options: Found no match for SIP option: sip-cc-01 (Please file bug report!) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:1686 parse_sip_options: Found SIP option: -replaces- [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:1692 parse_sip_options: Matched SIP option: replaces [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:1686 parse_sip_options: Found SIP option: -timer- [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:1692 parse_sip_options: Matched SIP option: timer [Mar 10 12:07:37] Sending to 192.168.1.85 : 5060 (no NAT) [Mar 10 12:07:37] Found RTP audio format 0 [Mar 10 12:07:37] Found RTP audio format 101 [Mar 10 12:07:37] Peer audio RTP is at port 0.0.0.0:30016 [Mar 10 12:07:37] Found description format PCMU for ID 0 [Mar 10 12:07:37] Found description format telephone-event for ID 101 [Mar 10 12:07:37] Got unsupported a:fmtp in SDP offer [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:5123 process_sdp: T38 state changed to 0 on channel SIP/523-087dd9a8 [Mar 10 12:07:37] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 10 12:07:37] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 10 12:07:37] Peer audio RTP is at port 0.0.0.0:30016 [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:5203 process_sdp: We're settling with these formats: 0x4 (ulaw) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:5210 process_sdp: We have an owner, now see if we need to change this call [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:13431 handle_request_invite: Got a SIP re-invite for call 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:13526 handle_request_invite: SIP/523-087dd9a8: This call is UP.... [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:6414 transmit_response_with_sdp: Setting framing from config on incoming call [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:6182 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: True [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:6183 add_sdp: ** Our prefcodec: 0x4 (ulaw) [Mar 10 12:07:37] Audio is at 24.123.23.170 port 10848 [Mar 10 12:07:37] Adding codec 0x4 (ulaw) to SDP [Mar 10 12:07:37] Adding non-codec 0x1 (telephone-event) to SDP [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:6314 add_sdp: -- Done with adding codecs to SDP [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:6359 add_sdp: Done building SDP. Settling with this capability: 0x4 (ulaw) [Mar 10 12:07:37] <--- Reliably Transmitting (no NAT) to 192.168.1.85:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKc128ad6bf7a12c24938883bffed31855b;received=192.168.1.85 From: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec To: ;tag=as628c2367 Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 388020 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 18522 18524 IN IP4 192.168.1.83 s=session c=IN IP4 192.168.1.83 t=0 0 m=audio 51000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #192 [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:16860 sip_set_rtp_peer: Sending reinvite on SIP 'a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83' - It's audio soon redirected to IP 24.123.23.170 [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:5637 reqprep: Strict routing enforced for session a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 [Mar 10 12:07:37] set_destination: Parsing for address/port to send to [Mar 10 12:07:37] set_destination: set destination to 192.168.1.83, port 5060 [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:6182 add_sdp: ** Our capability: 0xc (ulaw|alaw) Video flag: True [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:6183 add_sdp: ** Our prefcodec: 0x0 (nothing) [Mar 10 12:07:37] Audio is at 24.123.23.170 port 14996 [Mar 10 12:07:37] Adding codec 0x4 (ulaw) to SDP [Mar 10 12:07:37] Adding codec 0x8 (alaw) to SDP [Mar 10 12:07:37] Adding non-codec 0x1 (telephone-event) to SDP [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:6314 add_sdp: -- Done with adding codecs to SDP [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:6359 add_sdp: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:1619 initialize_initreq: Initializing already initialized SIP dialog a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 (presumably reinvite) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 0: INVITE sip:593@192.168.1.83:5060 SIP/2.0 (40) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK618c5b2d (58) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 2: From: 523 ;tag=as43feeb80 (53) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 3: To: 593 ;tag=48204f4e1c15803 (56) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 4: Contact: (32) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 5: Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 (54) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 6: CSeq: 103 INVITE (16) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 10: Supported: replaces (19) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 12: Content-Type: application/sdp (29) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 13: Content-Length: 266 (19) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 14: (0) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: o=root 18522 18524 IN IP4 24.123.23.170 (39) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: s=session (9) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: c=IN IP4 24.123.23.170 (22) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: m=audio 14996 RTP/AVP 0 8 101 (29) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-16 (15) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=silenceSupp:off - - - - (25) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=sendrecv (10) [Mar 10 12:07:37] Reliably Transmitting (no NAT) to 192.168.1.83:5060: INVITE sip:593@192.168.1.83:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK618c5b2d From: 523 ;tag=as43feeb80 To: 593 ;tag=48204f4e1c15803 Contact: Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 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: 266 v=0 o=root 18522 18524 IN IP4 24.123.23.170 s=session c=IN IP4 24.123.23.170 t=0 0 m=audio 14996 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 --- [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #193 [Mar 10 12:07:37] DEBUG[18591]: channel.c:2845 set_format: Set channel SIP/593-b77052f8 to write format slin [Mar 10 12:07:37] -- Started music on hold, class 'default', on channel 'SIP/593-b77052f8' [Mar 10 12:07:37] DEBUG[18591]: channel.c:1997 ast_settimeout: Scheduling timer at 160 sample intervals [Mar 10 12:07:37] DEBUG[18591]: rtp.c:2806 bridge_native_loop: Oooh, 'SIP/523-087dd9a8' changed end address to 0.0.0.0:0 (format 4) [Mar 10 12:07:37] DEBUG[18591]: rtp.c:2808 bridge_native_loop: Oooh, 'SIP/523-087dd9a8' changed end vaddress to 0.0.0.0:0 (format 4) [Mar 10 12:07:37] DEBUG[18591]: rtp.c:2810 bridge_native_loop: Oooh, 'SIP/523-087dd9a8' was 192.168.1.85:30016/(format 4) [Mar 10 12:07:37] DEBUG[18591]: rtp.c:2812 bridge_native_loop: Oooh, 'SIP/523-087dd9a8' was 0.0.0.0:0/(format 4) [Mar 10 12:07:37] DEBUG[18591]: rtp.c:2670 ast_rtp_write: Ooh, format changed from unknown to ulaw [Mar 10 12:07:37] DEBUG[18591]: rtp.c:2687 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Mar 10 12:07:37] <--- SIP read from 192.168.1.85:5060 ---> ACK sip:593@24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKtc9764d6fb2844025c9fc449ec57fe4cd CSeq: 388020 ACK To: ;tag=as628c2367 Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 From: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec User-Agent: Uniden SIP Phone p2 Ver BS4.77 <-------------> [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: ACK sip:593@24.123.23.170 SIP/2.0 (33) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKtc9764d6fb2844025c9fc449ec57fe4cd (82) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 388020 ACK (16) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: ;tag=as628c2367 (42) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 (55) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: From: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec (74) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: (0) [Mar 10 12:07:37] --- (7 headers 0 lines) --- [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received ACK (6) - Command in SIP ACK [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #192 [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '53c067b5491f609931bc7a0965b73fd3@24.123.23.170' of Response 388020: Match Not Found [Mar 10 12:07:37] <--- SIP read from 192.168.1.83:5060 ---> SIP/2.0 200 OK Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 CSeq: 103 INVITE From: 523 ;tag=as43feeb80 To: 593 ;tag=48204f4e1c15803 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK618c5b2d Content-Length: 266 Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE Content-Type: application/sdp Supported: replaces Contact: 593 User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 v=0 o=MxSIP 0 1143068287 IN IP4 192.168.1.83 s=SIP Call c=IN IP4 192.168.1.83 t=0 0 m=audio 51000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:on - - - - <-------------> [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: SIP/2.0 200 OK (14) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 (54) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 103 INVITE (16) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: From: 523 ;tag=as43feeb80 (53) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: To: 593 ;tag=48204f4e1c15803 (56) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK618c5b2d (58) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Content-Length: 266 (19) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE (55) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Content-Type: application/sdp (29) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Supported: replaces (19) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Contact: 593 (40) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 (42) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: (0) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: o=MxSIP 0 1143068287 IN IP4 192.168.1.83 (40) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: s=SIP Call (10) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: c=IN IP4 192.168.1.83 (21) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: m=audio 51000 RTP/AVP 0 8 101 (29) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-15 (15) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=sendrecv (10) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=silenceSupp:on - - - - (24) [Mar 10 12:07:37] --- (12 headers 13 lines) --- [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:2069 __sip_ack: Acked pending invite 103 [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #193 [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on 'a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83' of Request 103: Match Not Found [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:11619 handle_response_invite: SIP response 200 to RE-invite on outgoing call a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 [Mar 10 12:07:37] Found RTP audio format 0 [Mar 10 12:07:37] Found RTP audio format 8 [Mar 10 12:07:37] Found RTP audio format 101 [Mar 10 12:07:37] Peer audio RTP is at port 192.168.1.83:51000 [Mar 10 12:07:37] Found description format PCMU for ID 0 [Mar 10 12:07:37] Found description format PCMA for ID 8 [Mar 10 12:07:37] Found description format telephone-event for ID 101 [Mar 10 12:07:37] Got unsupported a:fmtp in SDP offer [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:5123 process_sdp: T38 state changed to 0 on channel SIP/593-b77052f8 [Mar 10 12:07:37] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) [Mar 10 12:07:37] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 10 12:07:37] Peer audio RTP is at port 192.168.1.83:51000 [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:5203 process_sdp: We're settling with these formats: 0xc (ulaw|alaw) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:5210 process_sdp: We have an owner, now see if we need to change this call [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:3001 update_call_counter: Updating call counter for incoming call [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:7903 build_route: build_route: Retaining previous route: [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:11752 handle_response_invite: Strange... The other side of the bridge does not have a udptl struct [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:11757 handle_response_invite: T38 state changed to 0 on channel SIP [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:11760 handle_response_invite: T38 state changed to 0 on channel SIP/593-b77052f8 [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:5637 reqprep: Strict routing enforced for session a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 [Mar 10 12:07:37] set_destination: Parsing for address/port to send to [Mar 10 12:07:37] set_destination: set destination to 192.168.1.83, port 5060 [Mar 10 12:07:37] Transmitting (no NAT) to 192.168.1.83:5060: ACK sip:593@192.168.1.83:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK34eb4bed From: 523 ;tag=as43feeb80 To: 593 ;tag=48204f4e1c15803 Contact: Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:16860 sip_set_rtp_peer: Sending reinvite on SIP '53c067b5491f609931bc7a0965b73fd3@24.123.23.170' - It's audio soon redirected to IP 192.168.1.83 [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:5637 reqprep: Strict routing enforced for session 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 [Mar 10 12:07:37] set_destination: Parsing for address/port to send to [Mar 10 12:07:37] set_destination: set destination to 192.168.1.85, port 5060 [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:6182 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: True [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:6183 add_sdp: ** Our prefcodec: 0x4 (ulaw) [Mar 10 12:07:37] Audio is at 24.123.23.170 port 10848 [Mar 10 12:07:37] Adding codec 0x4 (ulaw) to SDP [Mar 10 12:07:37] Adding non-codec 0x1 (telephone-event) to SDP [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:6314 add_sdp: -- Done with adding codecs to SDP [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:6359 add_sdp: Done building SDP. Settling with this capability: 0x4 (ulaw) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:1619 initialize_initreq: Initializing already initialized SIP dialog 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 (presumably reinvite) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 0: INVITE sip:523@192.168.1.85:5060 SIP/2.0 (40) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK57e2d3e9 (58) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 2: From: ;tag=as628c2367 (44) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 3: To: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec (72) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 4: Contact: (32) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 5: Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 (55) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 6: CSeq: 104 INVITE (16) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 10: Supported: replaces (19) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 12: Content-Type: application/sdp (29) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 13: Content-Length: 240 (19) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 14: (0) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: o=root 18522 18525 IN IP4 192.168.1.83 (38) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: s=session (9) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: c=IN IP4 192.168.1.83 (21) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: m=audio 51000 RTP/AVP 0 101 (27) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-16 (15) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=silenceSupp:off - - - - (25) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=recvonly (10) [Mar 10 12:07:37] Reliably Transmitting (no NAT) to 192.168.1.85:5060: INVITE sip:523@192.168.1.85:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK57e2d3e9 From: ;tag=as628c2367 To: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec Contact: Call-ID: 53c067b5491f609931bc7a0965b73fd3@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: 240 v=0 o=root 18522 18525 IN IP4 192.168.1.83 s=session c=IN IP4 192.168.1.83 t=0 0 m=audio 51000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly --- [Mar 10 12:07:37] DEBUG[18591]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #195 [Mar 10 12:07:37] <--- SIP read from 192.168.1.85:5060 ---> SIP/2.0 200 OK To: ;tag=fb7c7b32eac4985d4d8676d2196e3753 Contact: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK14f10e49 From: "551 551" ;tag=as59744546 Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 CSeq: 102 INVITE Content-Type: application/sdp Content-Length: 245 Supported: sip-cc, sip-cc-01, replaces, timer Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK v=0 o=- 18522 5561 IN IP4 192.168.1.85 s=session c=IN IP4 192.168.1.85 t=0 0 m=audio 30018 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 <-------------> [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: SIP/2.0 200 OK (14) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: To: ;tag=fb7c7b32eac4985d4d8676d2196e3753 (68) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Contact: (36) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK14f10e49 (58) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: "551 551" ;tag=as59744546 (54) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 (55) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: CSeq: 102 INVITE (16) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Content-Type: application/sdp (29) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Content-Length: 245 (19) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: (0) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: o=- 18522 5561 IN IP4 192.168.1.85 (34) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: s=session (9) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: c=IN IP4 192.168.1.85 (21) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: m=audio 30018 RTP/AVP 0 101 (27) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-15,16,17,18 (24) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=silenceSupp:off - - - - (25) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=sendrecv (10) [Mar 10 12:07:37] --- (11 headers 12 lines) --- [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:2069 __sip_ack: Acked pending invite 102 [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170' of Request 102: Match Not Found [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:11621 handle_response_invite: SIP response 200 to standard invite [Mar 10 12:07:37] Found RTP audio format 0 [Mar 10 12:07:37] Found RTP audio format 101 [Mar 10 12:07:37] Peer audio RTP is at port 192.168.1.85:30018 [Mar 10 12:07:37] Found description format PCMU for ID 0 [Mar 10 12:07:37] Found description format telephone-event for ID 101 [Mar 10 12:07:37] Got unsupported a:fmtp in SDP offer [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:5123 process_sdp: T38 state changed to 0 on channel SIP/523-087e2e08 [Mar 10 12:07:37] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 10 12:07:37] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 10 12:07:37] Peer audio RTP is at port 192.168.1.85:30018 [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:5203 process_sdp: We're settling with these formats: 0x4 (ulaw) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:5210 process_sdp: We have an owner, now see if we need to change this call [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:3001 update_call_counter: Updating call counter for outgoing call [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:7964 build_route: build_route: Contact hop: [Mar 10 12:07:37] list_route: hop: [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:5637 reqprep: Strict routing enforced for session 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 [Mar 10 12:07:37] set_destination: Parsing for address/port to send to [Mar 10 12:07:37] set_destination: set destination to 192.168.1.85, port 5060 [Mar 10 12:07:37] Transmitting (no NAT) to 192.168.1.85:5060: ACK sip:523@192.168.1.85:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5f030b2e From: "551 551" ;tag=as59744546 To: ;tag=fb7c7b32eac4985d4d8676d2196e3753 Contact: Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 10 12:07:37] -- Call on SIP/523-087e2e08 left from hold [Mar 10 12:07:37] DEBUG[18599]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/523-087e2e08 [Mar 10 12:07:37] -- SIP/523-087e2e08 answered SIP/551-b770d980 [Mar 10 12:07:37] DEBUG[18599]: rtp.c:1514 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/551-b770d980' with that of 'SIP/523-087e2e08' [Mar 10 12:07:37] DEBUG[18599]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/551-b770d980 [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:3461 sip_answer: SIP answering channel: SIP/551-b770d980 [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:6414 transmit_response_with_sdp: Setting framing from config on incoming call [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:6182 add_sdp: ** Our capability: 0xc (ulaw|alaw) Video flag: True [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:6183 add_sdp: ** Our prefcodec: 0x0 (nothing) [Mar 10 12:07:37] Audio is at 24.123.23.170 port 12970 [Mar 10 12:07:37] Adding codec 0x4 (ulaw) to SDP [Mar 10 12:07:37] Adding codec 0x8 (alaw) to SDP [Mar 10 12:07:37] Adding non-codec 0x1 (telephone-event) to SDP [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:6314 add_sdp: -- Done with adding codecs to SDP [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:6359 add_sdp: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Mar 10 12:07:37] <--- Reliably Transmitting (NAT) to 192.168.1.97:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.97:5060;branch=z9hG4bK1788404449;received=192.168.1.97;rport=5060 From: "Display Name" ;tag=2534858470 To: ;tag=as08598f69 Call-ID: 2919426183@192.168.1.97 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: 266 v=0 o=root 18522 18522 IN IP4 24.123.23.170 s=session c=IN IP4 24.123.23.170 t=0 0 m=audio 12970 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 <------------> [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #196 [Mar 10 12:07:37] -- Native bridging SIP/551-b770d980 and SIP/523-087e2e08 [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:16865 sip_set_rtp_peer: Deferring reinvite on SIP '2919426183@192.168.1.97' - It's audio will be redirected to IP 192.168.1.85 [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:16860 sip_set_rtp_peer: Sending reinvite on SIP '6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170' - It's audio soon redirected to IP 192.168.1.97 [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:5637 reqprep: Strict routing enforced for session 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 [Mar 10 12:07:37] set_destination: Parsing for address/port to send to [Mar 10 12:07:37] set_destination: set destination to 192.168.1.85, port 5060 [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:6182 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: True [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:6183 add_sdp: ** Our prefcodec: 0x4 (ulaw) [Mar 10 12:07:37] Audio is at 24.123.23.170 port 11786 [Mar 10 12:07:37] Adding codec 0x4 (ulaw) to SDP [Mar 10 12:07:37] Adding non-codec 0x1 (telephone-event) to SDP [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:6314 add_sdp: -- Done with adding codecs to SDP [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:6359 add_sdp: Done building SDP. Settling with this capability: 0x4 (ulaw) [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:1619 initialize_initreq: Initializing already initialized SIP dialog 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 (presumably reinvite) [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 0: INVITE sip:523@192.168.1.85:5060 SIP/2.0 (40) [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5de136bf (58) [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 2: From: "551 551" ;tag=as59744546 (54) [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 3: To: ;tag=fb7c7b32eac4985d4d8676d2196e3753 (68) [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 4: Contact: (32) [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 5: Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 (55) [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 6: CSeq: 103 INVITE (16) [Mar 10 12:07:37] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 523 [Mar 10 12:07:37] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 523 [Mar 10 12:07:37] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/523 - state 1 (Not in use) [Mar 10 12:07:37] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 551 [Mar 10 12:07:37] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 551 [Mar 10 12:07:37] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/551 - state 1 (Not in use) [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 10: Supported: replaces (19) [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 12: Content-Type: application/sdp (29) [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 13: Content-Length: 240 (19) [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 14: (0) [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:4603 parse_request: Line: o=root 18522 18523 IN IP4 192.168.1.97 (38) [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:4603 parse_request: Line: s=session (9) [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:4603 parse_request: Line: c=IN IP4 192.168.1.97 (21) [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:4603 parse_request: Line: m=audio 10010 RTP/AVP 0 101 (27) [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-16 (15) [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:4603 parse_request: Line: a=silenceSupp:off - - - - (25) [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:4603 parse_request: Line: a=sendrecv (10) [Mar 10 12:07:37] Reliably Transmitting (no NAT) to 192.168.1.85:5060: INVITE sip:523@192.168.1.85:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5de136bf From: "551 551" ;tag=as59744546 To: ;tag=fb7c7b32eac4985d4d8676d2196e3753 Contact: Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@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 18522 18523 IN IP4 192.168.1.97 s=session c=IN IP4 192.168.1.97 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=sendrecv --- [Mar 10 12:07:37] DEBUG[18599]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #197 [Mar 10 12:07:37] DEBUG[18604]: 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. [Mar 10 12:07:37] DEBUG[18605]: app_queue.c:546 changethread: Device 'SIP/551' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Mar 10 12:07:37] DEBUG[18599]: rtp.c:871 ast_rtcp_read: Got RTCP report of 64 bytes [Mar 10 12:07:37] DEBUG[18591]: channel.c:2332 __ast_read: Generator got voice, switching to phase locked mode [Mar 10 12:07:37] DEBUG[18591]: channel.c:1997 ast_settimeout: Scheduling timer at 0 sample intervals [Mar 10 12:07:37] <--- SIP read from 192.168.1.85:5060 ---> SIP/2.0 200 OK To: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec Contact: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK57e2d3e9 From: ;tag=as628c2367 Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 104 INVITE Content-Length: 242 Supported: sip-cc, sip-cc-01, replaces, timer Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK v=0 o=- 18522 204407 IN IP4 192.168.1.85 s=session c=IN IP4 0.0.0.0 t=0 0 m=audio 30016 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 <-------------> [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: SIP/2.0 200 OK (14) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: To: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec (72) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Contact: (36) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK57e2d3e9 (58) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=as628c2367 (44) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 (55) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: CSeq: 104 INVITE (16) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Content-Length: 242 (19) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: (0) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: o=- 18522 204407 IN IP4 192.168.1.85 (36) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: s=session (9) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: c=IN IP4 0.0.0.0 (16) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: m=audio 30016 RTP/AVP 0 101 (27) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-15,16,17,18 (24) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=silenceSupp:off - - - - (25) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=sendonly (10) [Mar 10 12:07:37] --- (10 headers 12 lines) --- [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:2069 __sip_ack: Acked pending invite 104 [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #195 [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '53c067b5491f609931bc7a0965b73fd3@24.123.23.170' of Request 104: Match Not Found [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:11619 handle_response_invite: SIP response 200 to RE-invite on outgoing call 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:3001 update_call_counter: Updating call counter for outgoing call [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:7903 build_route: build_route: Retaining previous route: [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:11752 handle_response_invite: Strange... The other side of the bridge does not have a udptl struct [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:11757 handle_response_invite: T38 state changed to 0 on channel SIP [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:11760 handle_response_invite: T38 state changed to 0 on channel SIP/523-087dd9a8 [Mar 10 12:07:37] DEBUG[18555]: chan_sip.c:5637 reqprep: Strict routing enforced for session 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 [Mar 10 12:07:37] set_destination: Parsing for address/port to send to [Mar 10 12:07:37] set_destination: set destination to 192.168.1.85, port 5060 [Mar 10 12:07:37] Transmitting (no NAT) to 192.168.1.85:5060: ACK sip:523@192.168.1.85:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK50c0d3d3 From: ;tag=as628c2367 To: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec Contact: Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 104 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 10 12:07:37] DEBUG[18599]: rtp.c:2670 ast_rtp_write: Ooh, format changed from unknown to ulaw [Mar 10 12:07:37] DEBUG[18599]: rtp.c:2687 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Mar 10 12:07:38] <--- SIP read from 192.168.1.97:5060 ---> ACK sip:523@24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.97:5060;rport;branch=z9hG4bK3517619738 From: "Display Name" ;tag=2534858470 To: ;tag=as08598f69 Call-ID: 2919426183@192.168.1.97 CSeq: 1001 ACK max-forwards: 70 user-agent: UTSTARCOM F1000/Device ID-0007ba26174b Content-Length: 0 <-------------> [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: ACK sip:523@24.123.23.170 SIP/2.0 (33) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.97:5060;rport;branch=z9hG4bK3517619738 (65) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: From: "Display Name" ;tag=2534858470 (59) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: ;tag=as08598f69 (42) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 2919426183@192.168.1.97 (32) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: CSeq: 1001 ACK (14) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: max-forwards: 70 (16) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: user-agent: UTSTARCOM F1000/Device ID-0007ba26174b (50) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Content-Length: 0 (17) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: (0) [Mar 10 12:07:38] --- (9 headers 0 lines) --- [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received ACK (6) - Command in SIP ACK [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #196 [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '2919426183@192.168.1.97' of Response 1001: Match Not Found [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:11602 check_pendings: Sending pending reinvite on '2919426183@192.168.1.97' [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:5637 reqprep: Strict routing enforced for session 2919426183@192.168.1.97 [Mar 10 12:07:38] set_destination: Parsing for address/port to send to [Mar 10 12:07:38] set_destination: set destination to 192.168.1.97, port 5060 [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:6182 add_sdp: ** Our capability: 0xc (ulaw|alaw) Video flag: True [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:6183 add_sdp: ** Our prefcodec: 0x0 (nothing) [Mar 10 12:07:38] Audio is at 24.123.23.170 port 12970 [Mar 10 12:07:38] Adding codec 0x4 (ulaw) to SDP [Mar 10 12:07:38] Adding codec 0x8 (alaw) to SDP [Mar 10 12:07:38] Adding non-codec 0x1 (telephone-event) to SDP [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:6314 add_sdp: -- Done with adding codecs to SDP [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:6359 add_sdp: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:1619 initialize_initreq: Initializing already initialized SIP dialog 2919426183@192.168.1.97 (presumably reinvite) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: INVITE sip:551@192.168.1.97:5060 SIP/2.0 (40) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK2598a417;rport (64) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: From: ;tag=as08598f69 (44) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: "Display Name" ;tag=2534858470 (57) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Contact: (32) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: Call-ID: 2919426183@192.168.1.97 (32) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: CSeq: 102 INVITE (16) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: replaces (19) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Content-Type: application/sdp (29) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: Content-Length: 264 (19) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 14: (0) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: o=root 18522 18523 IN IP4 192.168.1.85 (38) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: s=session (9) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: c=IN IP4 192.168.1.85 (21) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: m=audio 30018 RTP/AVP 0 8 101 (29) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-16 (15) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=silenceSupp:off - - - - (25) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=sendrecv (10) [Mar 10 12:07:38] Reliably Transmitting (NAT) to 192.168.1.97:5060: INVITE sip:551@192.168.1.97:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK2598a417;rport From: ;tag=as08598f69 To: "Display Name" ;tag=2534858470 Contact: Call-ID: 2919426183@192.168.1.97 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: 264 v=0 o=root 18522 18523 IN IP4 192.168.1.85 s=session c=IN IP4 192.168.1.85 t=0 0 m=audio 30018 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 --- [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #200 [Mar 10 12:07:38] <--- SIP read from 192.168.1.97:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK2598a417;rport=5060 From: ;tag=as08598f69 To: "Display Name" ;tag=2534858470 Call-ID: 2919426183@192.168.1.97 CSeq: 102 INVITE Contact: Content-Type: application/sdp Content-Length: 199 v=0 o=551 123456 654333 IN IP4 192.168.1.97 s=none c=IN IP4 192.168.1.97 t=0 0 m=audio 10010 RTP/AVP 0 101 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: SIP/2.0 200 OK (14) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK2598a417;rport=5060 (69) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: From: ;tag=as08598f69 (44) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: "Display Name" ;tag=2534858470 (57) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 2919426183@192.168.1.97 (32) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: CSeq: 102 INVITE (16) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: (36) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Content-Type: application/sdp (29) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Content-Length: 199 (21) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: (0) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: o=551 123456 654333 IN IP4 192.168.1.97 (39) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: s=none (6) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: c=IN IP4 192.168.1.97 (21) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: m=audio 10010 RTP/AVP 0 101 (27) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-16 (15) [Mar 10 12:07:38] --- (9 headers 10 lines) --- [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:2069 __sip_ack: Acked pending invite 102 [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #200 [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '2919426183@192.168.1.97' of Request 102: Match Not Found [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:11619 handle_response_invite: SIP response 200 to RE-invite on outgoing call 2919426183@192.168.1.97 [Mar 10 12:07:38] Found RTP audio format 0 [Mar 10 12:07:38] Found RTP audio format 101 [Mar 10 12:07:38] Peer audio RTP is at port 192.168.1.97:10010 [Mar 10 12:07:38] Found description format PCMU for ID 0 [Mar 10 12:07:38] Found description format telephone-event for ID 101 [Mar 10 12:07:38] Got unsupported a:fmtp in SDP offer [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:5123 process_sdp: T38 state changed to 0 on channel SIP/551-b770d980 [Mar 10 12:07:38] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 10 12:07:38] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 10 12:07:38] Peer audio RTP is at port 192.168.1.97:10010 [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:5203 process_sdp: We're settling with these formats: 0x4 (ulaw) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:5210 process_sdp: We have an owner, now see if we need to change this call [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:3001 update_call_counter: Updating call counter for incoming call [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:7964 build_route: build_route: Contact hop: [Mar 10 12:07:38] list_route: hop: [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:11752 handle_response_invite: Strange... The other side of the bridge does not have a udptl struct [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:11757 handle_response_invite: T38 state changed to 0 on channel SIP [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:11760 handle_response_invite: T38 state changed to 0 on channel SIP/551-b770d980 [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:5637 reqprep: Strict routing enforced for session 2919426183@192.168.1.97 [Mar 10 12:07:38] set_destination: Parsing for address/port to send to [Mar 10 12:07:38] set_destination: set destination to 192.168.1.97, port 5060 [Mar 10 12:07:38] Transmitting (NAT) to 192.168.1.97:5060: ACK sip:551@192.168.1.97:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK00d467e8;rport From: ;tag=as08598f69 To: "Display Name" ;tag=2534858470 Contact: Call-ID: 2919426183@192.168.1.97 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 10 12:07:38] <--- SIP read from 192.168.1.85:5060 ---> SIP/2.0 200 OK To: ;tag=fb7c7b32eac4985d4d8676d2196e3753 Contact: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5de136bf From: "551 551" ;tag=as59744546 Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 CSeq: 103 INVITE Content-Type: application/sdp Content-Length: 245 Supported: sip-cc, sip-cc-01, replaces, timer Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK v=0 o=- 18522 5562 IN IP4 192.168.1.85 s=session c=IN IP4 192.168.1.85 t=0 0 m=audio 30018 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 <-------------> [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: SIP/2.0 200 OK (14) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: To: ;tag=fb7c7b32eac4985d4d8676d2196e3753 (68) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Contact: (36) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5de136bf (58) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: "551 551" ;tag=as59744546 (54) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 (55) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: CSeq: 103 INVITE (16) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Content-Type: application/sdp (29) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Content-Length: 245 (19) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: (0) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: o=- 18522 5562 IN IP4 192.168.1.85 (34) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: s=session (9) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: c=IN IP4 192.168.1.85 (21) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: m=audio 30018 RTP/AVP 0 101 (27) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-15,16,17,18 (24) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=silenceSupp:off - - - - (25) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=sendrecv (10) [Mar 10 12:07:38] --- (11 headers 12 lines) --- [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:2069 __sip_ack: Acked pending invite 103 [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #197 [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170' of Request 103: Match Not Found [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:11619 handle_response_invite: SIP response 200 to RE-invite on outgoing call 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 [Mar 10 12:07:38] Found RTP audio format 0 [Mar 10 12:07:38] Found RTP audio format 101 [Mar 10 12:07:38] Peer audio RTP is at port 192.168.1.85:30018 [Mar 10 12:07:38] Found description format PCMU for ID 0 [Mar 10 12:07:38] Found description format telephone-event for ID 101 [Mar 10 12:07:38] Got unsupported a:fmtp in SDP offer [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:5123 process_sdp: T38 state changed to 0 on channel SIP/523-087e2e08 [Mar 10 12:07:38] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 10 12:07:38] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 10 12:07:38] Peer audio RTP is at port 192.168.1.85:30018 [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:5203 process_sdp: We're settling with these formats: 0x4 (ulaw) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:5210 process_sdp: We have an owner, now see if we need to change this call [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:3001 update_call_counter: Updating call counter for outgoing call [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:7903 build_route: build_route: Retaining previous route: [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:11752 handle_response_invite: Strange... The other side of the bridge does not have a udptl struct [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:11757 handle_response_invite: T38 state changed to 0 on channel SIP [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:11760 handle_response_invite: T38 state changed to 0 on channel SIP/523-087e2e08 [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:5637 reqprep: Strict routing enforced for session 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 [Mar 10 12:07:38] set_destination: Parsing for address/port to send to [Mar 10 12:07:38] set_destination: set destination to 192.168.1.85, port 5060 [Mar 10 12:07:38] Transmitting (no NAT) to 192.168.1.85:5060: ACK sip:523@192.168.1.85:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK4f3d6f38 From: "551 551" ;tag=as59744546 To: ;tag=fb7c7b32eac4985d4d8676d2196e3753 Contact: Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 10 12:07:38] DEBUG[18599]: rtp.c:871 ast_rtcp_read: Got RTCP report of 40 bytes [Mar 10 12:07:38] <--- 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=z9hG4bK469807b4 From: To: Call-ID: 000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165 CSeq: 11575 REGISTER User-Agent: CSCO/7 Contact: Content-Length: 0 Expires: 60 <-------------> [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK469807b4 (58) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: From: (40) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: (38) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165 (58) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: CSeq: 11575 REGISTER (20) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: User-Agent: CSCO/7 (18) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Contact: (37) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Content-Length: 0 (17) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Expires: 60 (11) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: (0) [Mar 10 12:07:38] --- (10 headers 0 lines) --- [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:4308 sip_alloc: Allocating new SIP dialog for 000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165 - REGISTER (No RTP) [Mar 10 12:07:38] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:07:38] Using latest REGISTER request as basis request [Mar 10 12:07:38] Sending to 192.168.1.165 : 5060 (no NAT) [Mar 10 12:07:38] <--- Transmitting (no NAT) to 192.168.1.165:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK469807b4;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165 CSeq: 11575 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:38] <--- Transmitting (no NAT) to 192.168.1.165:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK469807b4;received=192.168.1.165 From: To: ;tag=as1ab80b72 Call-ID: 000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165 CSeq: 11575 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7d53a1f8" Content-Length: 0 <------------> [Mar 10 12:07:38] Scheduling destruction of SIP dialog '000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165' in 32000 ms (Method: REGISTER) [Mar 10 12:07:39] <--- 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=z9hG4bK621dc29d From: To: Call-ID: 000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165 CSeq: 11576 REGISTER User-Agent: CSCO/7 Contact: Authorization: Digest username="514",realm="asterisk",uri="sip:24.123.23.170",response="d7d30a39d84dacb1023df61bb3411097",nonce="7d53a1f8",algorithm=MD5 Content-Length: 0 Expires: 60 <-------------> [Mar 10 12:07:39] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:07:39] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK621dc29d (58) [Mar 10 12:07:39] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: From: (40) [Mar 10 12:07:39] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: (38) [Mar 10 12:07:39] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165 (58) [Mar 10 12:07:39] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: CSeq: 11576 REGISTER (20) [Mar 10 12:07:39] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: User-Agent: CSCO/7 (18) [Mar 10 12:07:39] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Contact: (37) [Mar 10 12:07:39] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Authorization: Digest username="514",realm="asterisk",uri="sip:24.123.23.170",response="d7d30a39d84dacb1023df61bb3411097",nonce="7d53a1f8",algorithm=MD5 (152) [Mar 10 12:07:39] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:07:39] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Expires: 60 (11) [Mar 10 12:07:39] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: (0) [Mar 10 12:07:39] --- (11 headers 0 lines) --- [Mar 10 12:07:39] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:07:39] Using latest REGISTER request as basis request [Mar 10 12:07:39] Sending to 192.168.1.165 : 5060 (no NAT) [Mar 10 12:07:39] <--- 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=z9hG4bK621dc29d;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165 CSeq: 11576 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:39] <--- 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=z9hG4bK621dc29d;received=192.168.1.165 From: To: ;tag=as1ab80b72 Call-ID: 000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165 CSeq: 11576 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Sat, 10 Mar 2007 17:07:39 GMT Content-Length: 0 <------------> [Mar 10 12:07:39] DEBUG[18555]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/514 [Mar 10 12:07:39] Scheduling destruction of SIP dialog '000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165' in 32000 ms (Method: REGISTER) [Mar 10 12:07:39] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 514 [Mar 10 12:07:39] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 514 [Mar 10 12:07:39] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/514 - state 1 (Not in use) [Mar 10 12:07:39] DEBUG[18606]: 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. [Mar 10 12:07:40] <--- 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=z9hG4bK77177c37 From: To: Call-ID: 000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165 CSeq: 11574 REGISTER User-Agent: CSCO/7 Contact: Content-Length: 0 Expires: 60 <-------------> [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK77177c37 (58) [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: From: (40) [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: (38) [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165 (58) [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: CSeq: 11574 REGISTER (20) [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: User-Agent: CSCO/7 (18) [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Contact: (37) [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Content-Length: 0 (17) [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Expires: 60 (11) [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: (0) [Mar 10 12:07:40] --- (10 headers 0 lines) --- [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:4308 sip_alloc: Allocating new SIP dialog for 000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165 - REGISTER (No RTP) [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:07:40] Using latest REGISTER request as basis request [Mar 10 12:07:40] Sending to 192.168.1.165 : 5060 (no NAT) [Mar 10 12:07:40] <--- 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=z9hG4bK77177c37;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165 CSeq: 11574 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:40] <--- 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=z9hG4bK77177c37;received=192.168.1.165 From: To: ;tag=as128e290f Call-ID: 000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165 CSeq: 11574 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="36566566" Content-Length: 0 <------------> [Mar 10 12:07:40] Scheduling destruction of SIP dialog '000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165' in 32000 ms (Method: REGISTER) [Mar 10 12:07:40] <--- SIP read from 192.168.1.85:5060 ---> REGISTER sip:24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKo1a3eb8965a6dcc08b1e52a6440c7de1c Call-ID: 55ae664767e0@24.123.23.170 CSeq: 66119 REGISTER From: ;tag=729d48a95861feb0f4e6cdeb5164a04e 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="1548d602", algorithm=MD5, uri="sip:24.123.23.170", username="523", response="33da57efa606b1f7ff2652786cade458" <-------------> [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKo1a3eb8965a6dcc08b1e52a6440c7de1c (82) [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Call-ID: 55ae664767e0@24.123.23.170 (35) [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: CSeq: 66119 REGISTER (20) [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=729d48a95861feb0f4e6cdeb5164a04e (66) [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="1548d602", algorithm=MD5, uri="sip:24.123.23.170", username="523", response="33da57efa606b1f7ff2652786cade458" (167) [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:07:40] --- (13 headers 0 lines) --- [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:4308 sip_alloc: Allocating new SIP dialog for 55ae664767e0@24.123.23.170 - REGISTER (No RTP) [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:07:40] Using latest REGISTER request as basis request [Mar 10 12:07:40] Sending to 192.168.1.85 : 5060 (no NAT) [Mar 10 12:07:40] <--- Transmitting (no NAT) to 192.168.1.85:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKo1a3eb8965a6dcc08b1e52a6440c7de1c;received=192.168.1.85 From: ;tag=729d48a95861feb0f4e6cdeb5164a04e To: Call-ID: 55ae664767e0@24.123.23.170 CSeq: 66119 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:40] <--- Transmitting (no NAT) to 192.168.1.85:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKo1a3eb8965a6dcc08b1e52a6440c7de1c;received=192.168.1.85 From: ;tag=729d48a95861feb0f4e6cdeb5164a04e To: ;tag=as26a7f150 Call-ID: 55ae664767e0@24.123.23.170 CSeq: 66119 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1e82850a" Content-Length: 0 <------------> [Mar 10 12:07:40] Scheduling destruction of SIP dialog '55ae664767e0@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:07:40] <--- 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=z9hG4bK7ece397f From: To: Call-ID: 000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165 CSeq: 11575 REGISTER User-Agent: CSCO/7 Contact: Authorization: Digest username="513",realm="asterisk",uri="sip:24.123.23.170",response="408e16507f34190f701348f62911896b",nonce="36566566",algorithm=MD5 Content-Length: 0 Expires: 60 <-------------> [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK7ece397f (58) [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: From: (40) [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: (38) [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165 (58) [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: CSeq: 11575 REGISTER (20) [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: User-Agent: CSCO/7 (18) [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Contact: (37) [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Authorization: Digest username="513",realm="asterisk",uri="sip:24.123.23.170",response="408e16507f34190f701348f62911896b",nonce="36566566",algorithm=MD5 (152) [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Expires: 60 (11) [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: (0) [Mar 10 12:07:40] --- (11 headers 0 lines) --- [Mar 10 12:07:40] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:07:40] Using latest REGISTER request as basis request [Mar 10 12:07:40] Sending to 192.168.1.165 : 5060 (no NAT) [Mar 10 12:07:40] <--- 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=z9hG4bK7ece397f;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165 CSeq: 11575 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:40] <--- 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=z9hG4bK7ece397f;received=192.168.1.165 From: To: ;tag=as128e290f Call-ID: 000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165 CSeq: 11575 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Sat, 10 Mar 2007 17:07:40 GMT Content-Length: 0 <------------> [Mar 10 12:07:40] DEBUG[18555]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/513 [Mar 10 12:07:40] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 513 [Mar 10 12:07:40] Scheduling destruction of SIP dialog '000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165' in 32000 ms (Method: REGISTER) [Mar 10 12:07:40] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 513 [Mar 10 12:07:40] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/513 - state 1 (Not in use) [Mar 10 12:07:40] DEBUG[18607]: 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. [Mar 10 12:07:40] <--- SIP read from 192.168.1.85:5060 ---> REGISTER sip:24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKo721c6f9198eb8298f628d15e7edec759 CSeq: 66120 REGISTER Call-ID: 55ae664767e0@24.123.23.170 From: ;tag=729d48a95861feb0f4e6cdeb5164a04e 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="1e82850a", algorithm=MD5, uri="sip:24.123.23.170", username="523", response="944128956302c2380eaaaa2554bc7818" <-------------> [Mar 10 12:07:42] DEBUG[18591]: rtp.c:871 ast_rtcp_read: Got RTCP report of 64 bytes [Mar 10 12:07:44] DEBUG[18591]: rtp.c:2540 ast_rtp_raw_write: Difference is 16624, ms is 2098 [Mar 10 12:07:43] DEBUG[18528]: res_musiconhold.c:564 monmp3thread: Only wrote -1 of 640 bytes to pipe [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKo721c6f9198eb8298f628d15e7edec759 (82) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 66120 REGISTER (20) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Call-ID: 55ae664767e0@24.123.23.170 (35) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=729d48a95861feb0f4e6cdeb5164a04e (66) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="1e82850a", algorithm=MD5, uri="sip:24.123.23.170", username="523", response="944128956302c2380eaaaa2554bc7818" (167) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:07:44] --- (13 headers 0 lines) --- [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:07:44] Using latest REGISTER request as basis request [Mar 10 12:07:44] Sending to 192.168.1.85 : 5060 (no NAT) [Mar 10 12:07:44] <--- 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=z9hG4bKo721c6f9198eb8298f628d15e7edec759;received=192.168.1.85 From: ;tag=729d48a95861feb0f4e6cdeb5164a04e To: Call-ID: 55ae664767e0@24.123.23.170 CSeq: 66120 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:44] <--- 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=z9hG4bKo721c6f9198eb8298f628d15e7edec759;received=192.168.1.85 From: ;tag=729d48a95861feb0f4e6cdeb5164a04e To: ;tag=as26a7f150 Call-ID: 55ae664767e0@24.123.23.170 CSeq: 66120 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Sat, 10 Mar 2007 17:07:44 GMT Content-Length: 0 <------------> [Mar 10 12:07:44] DEBUG[18555]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/523 [Mar 10 12:07:44] Scheduling destruction of SIP dialog '55ae664767e0@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4308 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: OPTIONS sip:sip.broadvoice.com SIP/2.0 (38) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK3450cf74;rport (64) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: From: "asterisk" ;tag=as75097df8 (60) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: (28) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Contact: (37) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: Call-ID: 0e762358451464155b13c07b7637b757@24.123.23.170 (55) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Date: Sat, 10 Mar 2007 17:07:44 GMT (35) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Supported: replaces (19) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Content-Length: 0 (17) [Mar 10 12:07:44] 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=z9hG4bK3450cf74;rport From: "asterisk" ;tag=as75097df8 To: Contact: Call-ID: 0e762358451464155b13c07b7637b757@24.123.23.170 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sat, 10 Mar 2007 17:07:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #210 [Mar 10 12:07:44] <--- 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=z9hG4bKo721c6f9198eb8298f628d15e7edec759 CSeq: 66120 REGISTER Call-ID: 55ae664767e0@24.123.23.170 From: ;tag=729d48a95861feb0f4e6cdeb5164a04e 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="1e82850a", algorithm=MD5, uri="sip:24.123.23.170", username="523", response="944128956302c2380eaaaa2554bc7818" <-------------> [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKo721c6f9198eb8298f628d15e7edec759 (82) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 66120 REGISTER (20) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Call-ID: 55ae664767e0@24.123.23.170 (35) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=729d48a95861feb0f4e6cdeb5164a04e (66) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:07:44] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 523 [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:07:44] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 523 [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:07:44] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/523 - state 1 (Not in use) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="1e82850a", algorithm=MD5, uri="sip:24.123.23.170", username="523", response="944128956302c2380eaaaa2554bc7818" (167) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:07:44] --- (13 headers 0 lines) --- [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 66120, ours 66120) [Mar 10 12:07:44] Using latest REGISTER request as basis request [Mar 10 12:07:44] Sending to 192.168.1.85 : 5060 (no NAT) [Mar 10 12:07:44] <--- 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=z9hG4bKo721c6f9198eb8298f628d15e7edec759;received=192.168.1.85 From: ;tag=729d48a95861feb0f4e6cdeb5164a04e To: Call-ID: 55ae664767e0@24.123.23.170 CSeq: 66120 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:44] <--- 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=z9hG4bKo721c6f9198eb8298f628d15e7edec759;received=192.168.1.85 From: ;tag=729d48a95861feb0f4e6cdeb5164a04e To: ;tag=as26a7f150 Call-ID: 55ae664767e0@24.123.23.170 CSeq: 66120 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Sat, 10 Mar 2007 17:07:44 GMT Content-Length: 0 <------------> [Mar 10 12:07:44] DEBUG[18555]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/523 [Mar 10 12:07:44] Scheduling destruction of SIP dialog '55ae664767e0@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:07:44] <--- SIP read from 68.58.36.157:5060 ---> <-------------> [Mar 10 12:07:44] --- (0 headers 0 lines) Nat keepalive --- [Mar 10 12:07:44] <--- SIP read from 192.168.1.86:5062 ---> <-------------> [Mar 10 12:07:44] --- (0 headers 0 lines) Nat keepalive --- [Mar 10 12:07:44] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 523 [Mar 10 12:07:44] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 523 [Mar 10 12:07:44] <--- 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=z9hG4bKo721c6f9198eb8298f628d15e7edec759 CSeq: 66120 REGISTER Call-ID: 55ae664767e0@24.123.23.170 From: ;tag=729d48a95861feb0f4e6cdeb5164a04e 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="1e82850a", algorithm=MD5, uri="sip:24.123.23.170", username="523", response="944128956302c2380eaaaa2554bc7818" <-------------> [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKo721c6f9198eb8298f628d15e7edec759 (82) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 66120 REGISTER (20) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Call-ID: 55ae664767e0@24.123.23.170 (35) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=729d48a95861feb0f4e6cdeb5164a04e (66) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="1e82850a", algorithm=MD5, uri="sip:24.123.23.170", username="523", response="944128956302c2380eaaaa2554bc7818" (167) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:07:44] --- (13 headers 0 lines) --- [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 66120, ours 66120) [Mar 10 12:07:44] Using latest REGISTER request as basis request [Mar 10 12:07:44] Sending to 192.168.1.85 : 5060 (no NAT) [Mar 10 12:07:44] <--- 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=z9hG4bKo721c6f9198eb8298f628d15e7edec759;received=192.168.1.85 From: ;tag=729d48a95861feb0f4e6cdeb5164a04e To: Call-ID: 55ae664767e0@24.123.23.170 CSeq: 66120 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:44] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/523 - state 1 (Not in use) [Mar 10 12:07:44] DEBUG[18608]: app_queue.c:546 changethread: [Mar 10 12:07:44] <--- 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=z9hG4bKo721c6f9198eb8298f628d15e7edec759;received=192.168.1.85 From: ;tag=729d48a95861feb0f4e6cdeb5164a04e To: ;tag=as26a7f150 Call-ID: 55ae664767e0@24.123.23.170 CSeq: 66120 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Sat, 10 Mar 2007 17:07:44 GMT Content-Length: 0 <------------> Device 'SIP/523' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Mar 10 12:07:44] DEBUG[18555]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/523 [Mar 10 12:07:44] Scheduling destruction of SIP dialog '55ae664767e0@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:07:44] <--- SIP read from 192.168.1.85:5060 ---> INVITE sip:551@24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKec06c07c3079ca1c8c928ee81120eaffc Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 CSeq: 34085 INVITE From: 523 ;tag=fb7c7b32eac4985d4d8676d2196e3753 To: ;tag=as59744546 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: 240 v=0 o=- 18522 5563 IN IP4 192.168.1.85 s=session c=IN IP4 0.0.0.0 t=0 0 m=audio 30018 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 <-------------> [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: INVITE sip:551@24.123.23.170 SIP/2.0 (36) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKec06c07c3079ca1c8c928ee81120eaffc (82) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 (55) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: CSeq: 34085 INVITE (18) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: 523 ;tag=fb7c7b32eac4985d4d8676d2196e3753 (74) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: ;tag=as59744546 (42) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: (36) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Content-Type: application/sdp (29) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Max-Forwards: 70 (16) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Content-Length: 240 (19) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: o=- 18522 5563 IN IP4 192.168.1.85 (34) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: s=session (9) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: c=IN IP4 0.0.0.0 (16) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: m=audio 30018 RTP/AVP 0 101 (27) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-15,16,17,18 (24) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=silenceSupp:off - - - - (25) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=sendonly (10) [Mar 10 12:07:44] --- (13 headers 12 lines) --- [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:1678 parse_sip_options: Begin: parsing SIP "Supported: sip-cc, sip-cc-01, replaces, timer" [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:1686 parse_sip_options: Found SIP option: -sip-cc- [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:1700 parse_sip_options: Found no match for SIP option: sip-cc (Please file bug report!) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:1686 parse_sip_options: Found SIP option: -sip-cc-01- [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:1700 parse_sip_options: Found no match for SIP option: sip-cc-01 (Please file bug report!) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:1686 parse_sip_options: Found SIP option: -replaces- [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:1692 parse_sip_options: Matched SIP option: replaces [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:1686 parse_sip_options: Found SIP option: -timer- [Mar 10 12:07:44] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 523 [Mar 10 12:07:44] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 523 [Mar 10 12:07:44] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/523 - state 1 (Not in use) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:1692 parse_sip_options: Matched SIP option: timer [Mar 10 12:07:44] Sending to 192.168.1.85 : 5060 (no NAT) [Mar 10 12:07:44] Found RTP audio format 0 [Mar 10 12:07:44] Found RTP audio format 101 [Mar 10 12:07:44] Peer audio RTP is at port 0.0.0.0:30018 [Mar 10 12:07:44] Found description format PCMU for ID 0 [Mar 10 12:07:44] Found description format telephone-event for ID 101 [Mar 10 12:07:44] Got unsupported a:fmtp in SDP offer [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:5123 process_sdp: T38 state changed to 0 on channel SIP/523-087e2e08 [Mar 10 12:07:44] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 10 12:07:44] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 10 12:07:44] Peer audio RTP is at port 0.0.0.0:30018 [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:5203 process_sdp: We're settling with these formats: 0x4 (ulaw) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:5210 process_sdp: We have an owner, now see if we need to change this call [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:13431 handle_request_invite: Got a SIP re-invite for call 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:13526 handle_request_invite: SIP/523-087e2e08: This call is UP.... [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:6414 transmit_response_with_sdp: Setting framing from config on incoming call [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:6182 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: True [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:6183 add_sdp: ** Our prefcodec: 0x4 (ulaw) [Mar 10 12:07:44] Audio is at 24.123.23.170 port 11786 [Mar 10 12:07:44] Adding codec 0x4 (ulaw) to SDP [Mar 10 12:07:44] Adding non-codec 0x1 (telephone-event) to SDP [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:6314 add_sdp: -- Done with adding codecs to SDP [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:6359 add_sdp: Done building SDP. Settling with this capability: 0x4 (ulaw) [Mar 10 12:07:44] <--- Reliably Transmitting (no NAT) to 192.168.1.85:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKec06c07c3079ca1c8c928ee81120eaffc;received=192.168.1.85 From: 523 ;tag=fb7c7b32eac4985d4d8676d2196e3753 To: ;tag=as59744546 Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 CSeq: 34085 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 18522 18524 IN IP4 192.168.1.97 s=session c=IN IP4 192.168.1.97 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 <------------> [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #216 [Mar 10 12:07:44] DEBUG[18599]: chan_sip.c:16860 sip_set_rtp_peer: Sending reinvite on SIP '2919426183@192.168.1.97' - It's audio soon redirected to IP 24.123.23.170 [Mar 10 12:07:44] DEBUG[18599]: chan_sip.c:5637 reqprep: Strict routing enforced for session 2919426183@192.168.1.97 [Mar 10 12:07:44] set_destination: Parsing for address/port to send to [Mar 10 12:07:44] set_destination: set destination to 192.168.1.97, port 5060 [Mar 10 12:07:44] DEBUG[18599]: chan_sip.c:6182 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: True [Mar 10 12:07:44] DEBUG[18599]: chan_sip.c:6183 add_sdp: ** Our prefcodec: 0x0 (nothing) [Mar 10 12:07:44] Audio is at 24.123.23.170 port 12970 [Mar 10 12:07:44] Adding codec 0x4 (ulaw) to SDP [Mar 10 12:07:44] Adding non-codec 0x1 (telephone-event) to SDP [Mar 10 12:07:44] DEBUG[18599]: chan_sip.c:6314 add_sdp: -- Done with adding codecs to SDP [Mar 10 12:07:44] DEBUG[18599]: chan_sip.c:6359 add_sdp: Done building SDP. Settling with this capability: 0x4 (ulaw) [Mar 10 12:07:44] DEBUG[18599]: chan_sip.c:1619 initialize_initreq: Initializing already initialized SIP dialog 2919426183@192.168.1.97 (presumably reinvite) [Mar 10 12:07:44] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 0: INVITE sip:551@192.168.1.97:5060 SIP/2.0 (40) [Mar 10 12:07:44] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK4276259f;rport (64) [Mar 10 12:07:44] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 2: From: ;tag=as08598f69 (44) [Mar 10 12:07:44] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 3: To: "Display Name" ;tag=2534858470 (57) [Mar 10 12:07:44] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 4: Contact: (32) [Mar 10 12:07:44] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 5: Call-ID: 2919426183@192.168.1.97 (32) [Mar 10 12:07:44] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 6: CSeq: 103 INVITE (16) [Mar 10 12:07:44] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Mar 10 12:07:44] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:07:44] DEBUG[18609]: 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. [Mar 10 12:07:44] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Mar 10 12:07:44] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 10: Supported: replaces (19) [Mar 10 12:07:44] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Mar 10 12:07:44] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 12: Content-Type: application/sdp (29) [Mar 10 12:07:44] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 13: Content-Length: 242 (19) [Mar 10 12:07:44] DEBUG[18599]: chan_sip.c:4571 parse_request: Header 14: (0) [Mar 10 12:07:44] DEBUG[18599]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:07:44] DEBUG[18599]: chan_sip.c:4603 parse_request: Line: o=root 18522 18524 IN IP4 24.123.23.170 (39) [Mar 10 12:07:44] DEBUG[18599]: chan_sip.c:4603 parse_request: Line: s=session (9) [Mar 10 12:07:44] DEBUG[18599]: chan_sip.c:4603 parse_request: Line: c=IN IP4 24.123.23.170 (22) [Mar 10 12:07:44] DEBUG[18599]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:07:44] DEBUG[18599]: chan_sip.c:4603 parse_request: Line: m=audio 12970 RTP/AVP 0 101 (27) [Mar 10 12:07:44] DEBUG[18599]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:07:44] DEBUG[18599]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:07:44] DEBUG[18599]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-16 (15) [Mar 10 12:07:44] DEBUG[18599]: chan_sip.c:4603 parse_request: Line: a=silenceSupp:off - - - - (25) [Mar 10 12:07:44] DEBUG[18599]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:07:44] DEBUG[18599]: chan_sip.c:4603 parse_request: Line: a=sendrecv (10) [Mar 10 12:07:44] Reliably Transmitting (NAT) to 192.168.1.97:5060: INVITE sip:551@192.168.1.97:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK4276259f;rport From: ;tag=as08598f69 To: "Display Name" ;tag=2534858470 Contact: Call-ID: 2919426183@192.168.1.97 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 18522 18524 IN IP4 24.123.23.170 s=session c=IN IP4 24.123.23.170 t=0 0 m=audio 12970 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 --- [Mar 10 12:07:44] DEBUG[18599]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #217 [Mar 10 12:07:44] <--- 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=z9hG4bK0b86bc03 From: To: Call-ID: 000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165 CSeq: 11595 REGISTER User-Agent: CSCO/7 Contact: Content-Length: 0 Expires: 60 <-------------> [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK0b86bc03 (58) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: From: (40) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: (38) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165 (58) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: CSeq: 11595 REGISTER (20) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: User-Agent: CSCO/7 (18) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Contact: (37) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Content-Length: 0 (17) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Expires: 60 (11) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: (0) [Mar 10 12:07:44] --- (10 headers 0 lines) --- [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4308 sip_alloc: Allocating new SIP dialog for 000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165 - REGISTER (No RTP) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:07:44] Using latest REGISTER request as basis request [Mar 10 12:07:44] Sending to 192.168.1.165 : 5060 (no NAT) [Mar 10 12:07:44] <--- 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=z9hG4bK0b86bc03;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165 CSeq: 11595 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:44] <--- 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=z9hG4bK0b86bc03;received=192.168.1.165 From: To: ;tag=as601fb678 Call-ID: 000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165 CSeq: 11595 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="71e86f1c" Content-Length: 0 <------------> [Mar 10 12:07:44] Scheduling destruction of SIP dialog '000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165' in 32000 ms (Method: REGISTER) [Mar 10 12:07:44] <--- SIP read from 192.168.1.85:5060 ---> INVITE sip:551@24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKec06c07c3079ca1c8c928ee81120eaffc Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 CSeq: 34085 INVITE From: 523 ;tag=fb7c7b32eac4985d4d8676d2196e3753 To: ;tag=as59744546 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: 240 v=0 o=- 18522 5563 IN IP4 192.168.1.85 s=session c=IN IP4 0.0.0.0 t=0 0 m=audio 30018 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 <-------------> [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: INVITE sip:551@24.123.23.170 SIP/2.0 (36) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKec06c07c3079ca1c8c928ee81120eaffc (82) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 (55) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: CSeq: 34085 INVITE (18) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: 523 ;tag=fb7c7b32eac4985d4d8676d2196e3753 (74) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: ;tag=as59744546 (42) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: (36) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Content-Type: application/sdp (29) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Max-Forwards: 70 (16) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Content-Length: 240 (19) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: o=- 18522 5563 IN IP4 192.168.1.85 (34) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: s=session (9) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: c=IN IP4 0.0.0.0 (16) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: m=audio 30018 RTP/AVP 0 101 (27) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-15,16,17,18 (24) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=silenceSupp:off - - - - (25) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=sendonly (10) [Mar 10 12:07:44] --- (13 headers 12 lines) --- [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (INVITE Seqno 34085, ours 34085) [Mar 10 12:07:44] Ignoring this INVITE request [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:13433 handle_request_invite: Got a SIP re-transmit of INVITE for call 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:13526 handle_request_invite: SIP/523-087e2e08: This call is UP.... [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:6414 transmit_response_with_sdp: Setting framing from config on incoming call [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:6182 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: True [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:6183 add_sdp: ** Our prefcodec: 0x4 (ulaw) [Mar 10 12:07:44] Audio is at 24.123.23.170 port 11786 [Mar 10 12:07:44] Adding codec 0x4 (ulaw) to SDP [Mar 10 12:07:44] Adding non-codec 0x1 (telephone-event) to SDP [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:6314 add_sdp: -- Done with adding codecs to SDP [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:6359 add_sdp: Done building SDP. Settling with this capability: 0x4 (ulaw) [Mar 10 12:07:44] <--- Reliably Transmitting (no NAT) to 192.168.1.85:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKec06c07c3079ca1c8c928ee81120eaffc;received=192.168.1.85 From: 523 ;tag=fb7c7b32eac4985d4d8676d2196e3753 To: ;tag=as59744546 Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 CSeq: 34085 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 18522 18525 IN IP4 192.168.1.97 s=session c=IN IP4 192.168.1.97 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 <------------> [Mar 10 12:07:44] DEBUG[18610]: 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. [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #219 [Mar 10 12:07:44] DEBUG[18599]: channel.c:2845 set_format: Set channel SIP/551-b770d980 to write format slin [Mar 10 12:07:44] -- Started music on hold, class 'default', on channel 'SIP/551-b770d980' [Mar 10 12:07:44] DEBUG[18599]: channel.c:1997 ast_settimeout: Scheduling timer at 160 sample intervals [Mar 10 12:07:44] DEBUG[18599]: rtp.c:2806 bridge_native_loop: Oooh, 'SIP/523-087e2e08' changed end address to 0.0.0.0:0 (format 4) [Mar 10 12:07:44] <--- 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=z9hG4bK0b86bc03 From: To: Call-ID: 000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165 CSeq: 11595 REGISTER User-Agent: CSCO/7 Contact: Content-Length: 0 Expires: 60 <-------------> [Mar 10 12:07:44] DEBUG[18599]: rtp.c:2808 bridge_native_loop: Oooh, 'SIP/523-087e2e08' changed end vaddress to 0.0.0.0:0 (format 4) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:07:44] DEBUG[18599]: rtp.c:2810 bridge_native_loop: Oooh, 'SIP/523-087e2e08' was 192.168.1.85:30018/(format 4) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK0b86bc03 (58) [Mar 10 12:07:44] DEBUG[18599]: rtp.c:2812 bridge_native_loop: Oooh, 'SIP/523-087e2e08' was 0.0.0.0:0/(format 4) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: From: (40) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: (38) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165 (58) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: CSeq: 11595 REGISTER (20) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: User-Agent: CSCO/7 (18) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Contact: (37) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Content-Length: 0 (17) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Expires: 60 (11) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: (0) [Mar 10 12:07:44] --- (10 headers 0 lines) --- [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 11595, ours 11595) [Mar 10 12:07:44] Using latest REGISTER request as basis request [Mar 10 12:07:44] Sending to 192.168.1.165 : 5060 (no NAT) [Mar 10 12:07:44] <--- 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=z9hG4bK0b86bc03;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165 CSeq: 11595 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:44] <--- 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=z9hG4bK0b86bc03;received=192.168.1.165 From: To: ;tag=as601fb678 Call-ID: 000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165 CSeq: 11595 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="71e86f1c" Content-Length: 0 <------------> [Mar 10 12:07:44] Scheduling destruction of SIP dialog '000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165' in 32000 ms (Method: REGISTER) [Mar 10 12:07:44] <--- 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=z9hG4bKo721c6f9198eb8298f628d15e7edec759 CSeq: 66120 REGISTER Call-ID: 55ae664767e0@24.123.23.170 From: ;tag=729d48a95861feb0f4e6cdeb5164a04e 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="1e82850a", algorithm=MD5, uri="sip:24.123.23.170", username="523", response="944128956302c2380eaaaa2554bc7818" <-------------> [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKo721c6f9198eb8298f628d15e7edec759 (82) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 66120 REGISTER (20) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Call-ID: 55ae664767e0@24.123.23.170 (35) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=729d48a95861feb0f4e6cdeb5164a04e (66) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="1e82850a", algorithm=MD5, uri="sip:24.123.23.170", username="523", response="944128956302c2380eaaaa2554bc7818" (167) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:07:44] --- (13 headers 0 lines) --- [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 66120, ours 66120) [Mar 10 12:07:44] Using latest REGISTER request as basis request [Mar 10 12:07:44] Sending to 192.168.1.85 : 5060 (no NAT) [Mar 10 12:07:44] <--- 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=z9hG4bKo721c6f9198eb8298f628d15e7edec759;received=192.168.1.85 From: ;tag=729d48a95861feb0f4e6cdeb5164a04e To: Call-ID: 55ae664767e0@24.123.23.170 CSeq: 66120 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:44] <--- 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=z9hG4bKo721c6f9198eb8298f628d15e7edec759;received=192.168.1.85 From: ;tag=729d48a95861feb0f4e6cdeb5164a04e To: ;tag=as26a7f150 Call-ID: 55ae664767e0@24.123.23.170 CSeq: 66120 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Sat, 10 Mar 2007 17:07:44 GMT Content-Length: 0 <------------> [Mar 10 12:07:44] DEBUG[18555]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/523 [Mar 10 12:07:44] Scheduling destruction of SIP dialog '55ae664767e0@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:07:44] <--- SIP read from 192.168.1.85:5060 ---> INVITE sip:551@24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKec06c07c3079ca1c8c928ee81120eaffc Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 CSeq: 34085 INVITE From: 523 ;tag=fb7c7b32eac4985d4d8676d2196e3753 To: ;tag=as59744546 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: 240 v=0 o=- 18522 5563 IN IP4 192.168.1.85 s=session c=IN IP4 0.0.0.0 t=0 0 m=audio 30018 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 <-------------> [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: INVITE sip:551@24.123.23.170 SIP/2.0 (36) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKec06c07c3079ca1c8c928ee81120eaffc (82) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 (55) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: CSeq: 34085 INVITE (18) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: 523 ;tag=fb7c7b32eac4985d4d8676d2196e3753 (74) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: ;tag=as59744546 (42) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: (36) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Content-Type: application/sdp (29) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Max-Forwards: 70 (16) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Content-Length: 240 (19) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: o=- 18522 5563 IN IP4 192.168.1.85 (34) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: s=session (9) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: c=IN IP4 0.0.0.0 (16) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: m=audio 30018 RTP/AVP 0 101 (27) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-15,16,17,18 (24) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=silenceSupp:off - - - - (25) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=sendonly (10) [Mar 10 12:07:44] --- (13 headers 12 lines) --- [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (INVITE Seqno 34085, ours 34085) [Mar 10 12:07:44] Ignoring this INVITE request [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:13433 handle_request_invite: Got a SIP re-transmit of INVITE for call 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:13526 handle_request_invite: SIP/523-087e2e08: This call is UP.... [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:6414 transmit_response_with_sdp: Setting framing from config on incoming call [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:6182 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: True [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:6183 add_sdp: ** Our prefcodec: 0x4 (ulaw) [Mar 10 12:07:44] Audio is at 24.123.23.170 port 11786 [Mar 10 12:07:44] Adding codec 0x4 (ulaw) to SDP [Mar 10 12:07:44] Adding non-codec 0x1 (telephone-event) to SDP [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:6314 add_sdp: -- Done with adding codecs to SDP [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:6359 add_sdp: Done building SDP. Settling with this capability: 0x4 (ulaw) [Mar 10 12:07:44] <--- Reliably Transmitting (no NAT) to 192.168.1.85:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKec06c07c3079ca1c8c928ee81120eaffc;received=192.168.1.85 From: 523 ;tag=fb7c7b32eac4985d4d8676d2196e3753 To: ;tag=as59744546 Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 CSeq: 34085 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 18522 18526 IN IP4 192.168.1.97 s=session c=IN IP4 192.168.1.97 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 <------------> [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #223 [Mar 10 12:07:44] <--- 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=z9hG4bK26bd0c9d From: To: Call-ID: 000ff78d-ebb20007-4ad25281-15520cf9@192.168.1.165 CSeq: 11601 REGISTER User-Agent: CSCO/7 Contact: Content-Length: 0 Expires: 60 <-------------> [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK26bd0c9d (58) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: From: (40) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: (38) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 000ff78d-ebb20007-4ad25281-15520cf9@192.168.1.165 (58) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: CSeq: 11601 REGISTER (20) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: User-Agent: CSCO/7 (18) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Contact: (37) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Content-Length: 0 (17) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Expires: 60 (11) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: (0) [Mar 10 12:07:44] --- (10 headers 0 lines) --- [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4308 sip_alloc: Allocating new SIP dialog for 000ff78d-ebb20007-4ad25281-15520cf9@192.168.1.165 - REGISTER (No RTP) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:07:44] Using latest REGISTER request as basis request [Mar 10 12:07:44] Sending to 192.168.1.165 : 5060 (no NAT) [Mar 10 12:07:44] <--- 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=z9hG4bK26bd0c9d;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20007-4ad25281-15520cf9@192.168.1.165 CSeq: 11601 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:44] <--- 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=z9hG4bK26bd0c9d;received=192.168.1.165 From: To: ;tag=as0befcdde Call-ID: 000ff78d-ebb20007-4ad25281-15520cf9@192.168.1.165 CSeq: 11601 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="52145cc9" Content-Length: 0 <------------> [Mar 10 12:07:44] Scheduling destruction of SIP dialog '000ff78d-ebb20007-4ad25281-15520cf9@192.168.1.165' in 32000 ms (Method: REGISTER) [Mar 10 12:07:44] <--- 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=z9hG4bK0b86bc03 From: To: Call-ID: 000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165 CSeq: 11595 REGISTER User-Agent: CSCO/7 Contact: Content-Length: 0 Expires: 60 <-------------> [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK0b86bc03 (58) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: From: (40) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: (38) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165 (58) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: CSeq: 11595 REGISTER (20) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: User-Agent: CSCO/7 (18) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Contact: (37) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Content-Length: 0 (17) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Expires: 60 (11) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: (0) [Mar 10 12:07:44] --- (10 headers 0 lines) --- [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 11595, ours 11595) [Mar 10 12:07:44] Using latest REGISTER request as basis request [Mar 10 12:07:44] Sending to 192.168.1.165 : 5060 (no NAT) [Mar 10 12:07:44] <--- 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=z9hG4bK0b86bc03;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165 CSeq: 11595 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:44] <--- 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=z9hG4bK0b86bc03;received=192.168.1.165 From: To: ;tag=as601fb678 Call-ID: 000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165 CSeq: 11595 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="71e86f1c" Content-Length: 0 <------------> [Mar 10 12:07:44] Scheduling destruction of SIP dialog '000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165' in 32000 ms (Method: REGISTER) [Mar 10 12:07:44] <--- SIP read from 192.168.1.97:5060 ---> BYE sip:523@24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.97:5060;rport;branch=z9hG4bK1117930075 From: "Display Name" ;tag=2534858470 To: ;tag=as08598f69 Call-ID: 2919426183@192.168.1.97 CSeq: 1002 BYE max-forwards: 70 user-agent: UTSTARCOM F1000/Device ID-0007ba26174b Content-Length: 0 <-------------> [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: BYE sip:523@24.123.23.170 SIP/2.0 (33) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.97:5060;rport;branch=z9hG4bK1117930075 (65) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: From: "Display Name" ;tag=2534858470 (59) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: ;tag=as08598f69 (42) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 2919426183@192.168.1.97 (32) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: CSeq: 1002 BYE (14) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: max-forwards: 70 (16) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: user-agent: UTSTARCOM F1000/Device ID-0007ba26174b (50) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Content-Length: 0 (17) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: (0) [Mar 10 12:07:44] --- (9 headers 0 lines) --- [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received BYE (8) - Command in SIP BYE [Mar 10 12:07:44] Sending to 192.168.1.97 : 5060 (NAT) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:1631 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 2919426183@192.168.1.97 [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:14167 handle_request_bye: Received bye, issuing owner hangup [Mar 10 12:07:44] <--- Transmitting (NAT) to 192.168.1.97:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.97:5060;branch=z9hG4bK1117930075;received=192.168.1.97;rport=5060 From: "Display Name" ;tag=2534858470 To: ;tag=as08598f69 Call-ID: 2919426183@192.168.1.97 CSeq: 1002 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:44] DEBUG[18599]: channel.c:2845 set_format: Set channel SIP/551-b770d980 to write format ulaw [Mar 10 12:07:44] -- Stopped music on hold on SIP/551-b770d980 [Mar 10 12:07:44] DEBUG[18599]: channel.c:1997 ast_settimeout: Scheduling timer at 0 sample intervals [Mar 10 12:07:44] DEBUG[18599]: rtp.c:2855 bridge_native_loop: Oooh, got a hangup [Mar 10 12:07:44] DEBUG[18599]: chan_sip.c:16865 sip_set_rtp_peer: Deferring reinvite on SIP '6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170' - It's audio will be redirected to IP 24.123.23.170 [Mar 10 12:07:44] DEBUG[18599]: channel.c:4048 ast_channel_bridge: Returning from native bridge, channels: SIP/551-b770d980, SIP/523-087e2e08 [Mar 10 12:07:44] DEBUG[18599]: channel.c:1693 ast_hangup: Hanging up channel 'SIP/523-087e2e08' [Mar 10 12:07:44] DEBUG[18599]: chan_sip.c:3310 sip_hangup: Hangup call SIP/523-087e2e08, SIP callid 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170) [Mar 10 12:07:44] <--- 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=z9hG4bK26bd0c9d From: To: Call-ID: 000ff78d-ebb20007-4ad25281-15520cf9@192.168.1.165 CSeq: 11601 REGISTER User-Agent: CSCO/7 Contact: Content-Length: 0 Expires: 60 <-------------> [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK26bd0c9d (58) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: From: (40) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: (38) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 000ff78d-ebb20007-4ad25281-15520cf9@192.168.1.165 (58) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: CSeq: 11601 REGISTER (20) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: User-Agent: CSCO/7 (18) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Contact: (37) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Content-Length: 0 (17) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Expires: 60 (11) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: (0) [Mar 10 12:07:44] --- (10 headers 0 lines) --- [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 11601, ours 11601) [Mar 10 12:07:44] Using latest REGISTER request as basis request [Mar 10 12:07:44] Sending to 192.168.1.165 : 5060 (no NAT) [Mar 10 12:07:44] <--- 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=z9hG4bK26bd0c9d;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20007-4ad25281-15520cf9@192.168.1.165 CSeq: 11601 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:44] <--- 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=z9hG4bK26bd0c9d;received=192.168.1.165 From: To: ;tag=as0befcdde Call-ID: 000ff78d-ebb20007-4ad25281-15520cf9@192.168.1.165 CSeq: 11601 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="52145cc9" Content-Length: 0 <------------> [Mar 10 12:07:44] Scheduling destruction of SIP dialog '000ff78d-ebb20007-4ad25281-15520cf9@192.168.1.165' in 32000 ms (Method: REGISTER) [Mar 10 12:07:44] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 523 [Mar 10 12:07:44] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 523 [Mar 10 12:07:44] Scheduling destruction of SIP dialog '6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170' in 32000 ms (Method: INVITE) [Mar 10 12:07:44] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/523 - state 1 (Not in use) [Mar 10 12:07:44] DEBUG[18599]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/523-087e2e08 [Mar 10 12:07:44] DEBUG[18599]: rtp.c:1474 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Mar 10 12:07:44] DEBUG[18599]: app_dial.c:1670 dial_exec_full: Exiting with DIALSTATUS=ANSWER. [Mar 10 12:07:44] DEBUG[18599]: pbx.c:2389 __ast_pbx_run: Spawn extension (smvoice-sip,523,6) exited non-zero on 'SIP/551-b770d980' [Mar 10 12:07:44] == Spawn extension (smvoice-sip, 523, 6) exited non-zero on 'SIP/551-b770d980' [Mar 10 12:07:44] DEBUG[18599]: pbx.c:1644 pbx_substitute_variables_helper_full: Function result is '"551 551" <551>' [Mar 10 12:07:44] DEBUG[18599]: pbx.c:1644 pbx_substitute_variables_helper_full: Function result is '551' [Mar 10 12:07:44] DEBUG[18599]: pbx.c:1644 pbx_substitute_variables_helper_full: Function result is '523' [Mar 10 12:07:44] DEBUG[18599]: pbx.c:1644 pbx_substitute_variables_helper_full: Function result is 'smvoice-sip' [Mar 10 12:07:44] DEBUG[18599]: pbx.c:1644 pbx_substitute_variables_helper_full: Function result is 'SIP/551-b770d980' [Mar 10 12:07:44] DEBUG[18599]: pbx.c:1644 pbx_substitute_variables_helper_full: Function result is 'SIP/523-087e2e08' [Mar 10 12:07:44] DEBUG[18599]: pbx.c:1644 pbx_substitute_variables_helper_full: Function result is 'Dial' [Mar 10 12:07:44] DEBUG[18599]: pbx.c:1644 pbx_substitute_variables_helper_full: Function result is 'SIP/523|20|' [Mar 10 12:07:44] DEBUG[18599]: pbx.c:1644 pbx_substitute_variables_helper_full: Function result is '2007-03-10 12:07:36' [Mar 10 12:07:44] DEBUG[18599]: pbx.c:1644 pbx_substitute_variables_helper_full: Function result is '2007-03-10 12:07:37' [Mar 10 12:07:44] DEBUG[18599]: pbx.c:1644 pbx_substitute_variables_helper_full: Function result is '2007-03-10 12:07:44' [Mar 10 12:07:44] DEBUG[18599]: pbx.c:1644 pbx_substitute_variables_helper_full: Function result is '8' [Mar 10 12:07:44] DEBUG[18599]: pbx.c:1644 pbx_substitute_variables_helper_full: Function result is '7' [Mar 10 12:07:44] DEBUG[18599]: pbx.c:1644 pbx_substitute_variables_helper_full: Function result is 'ANSWERED' [Mar 10 12:07:44] DEBUG[18599]: pbx.c:1644 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' [Mar 10 12:07:44] DEBUG[18599]: pbx.c:1644 pbx_substitute_variables_helper_full: Function result is '' [Mar 10 12:07:44] DEBUG[18599]: pbx.c:1644 pbx_substitute_variables_helper_full: Function result is '1173546456.2' [Mar 10 12:07:44] DEBUG[18599]: pbx.c:1644 pbx_substitute_variables_helper_full: Function result is '' [Mar 10 12:07:44] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 523 [Mar 10 12:07:44] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 523 [Mar 10 12:07:44] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/523 - state 1 (Not in use) [Mar 10 12:07:44] DEBUG[18611]: 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. [Mar 10 12:07:44] DEBUG[18612]: 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. [Mar 10 12:07:44] <--- SIP read from 147.135.12.128:5060 ---> SIP/2.0 200 OK Call-ID: 0e762358451464155b13c07b7637b757@24.123.23.170 CSeq: 102 OPTIONS From: "asterisk" ;tag=as75097df8 To: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK3450cf74 Supported: 100rel Allow: INVITE, BYE, ACK, OPTIONS, CANCEL, PRACK Accept: application/sdp Accept-Encoding: Accept-Language: en Content-Length: 0 <-------------> [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: SIP/2.0 200 OK (14) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Call-ID: 0e762358451464155b13c07b7637b757@24.123.23.170 (55) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 102 OPTIONS (17) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: From: "asterisk" ;tag=as75097df8 (60) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: To: (28) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK3450cf74 (58) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Supported: 100rel (17) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Allow: INVITE, BYE, ACK, OPTIONS, CANCEL, PRACK (47) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Accept: application/sdp (23) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Accept-Encoding: (17) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Accept-Language: en (19) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Content-Length: 0 (20) [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: (0) [Mar 10 12:07:44] --- (12 headers 0 lines) --- [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #210 [Mar 10 12:07:44] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '0e762358451464155b13c07b7637b757@24.123.23.170' of Request 102: Match Not Found [Mar 10 12:07:44] Really destroying SIP dialog '0e762358451464155b13c07b7637b757@24.123.23.170' Method: OPTIONS [Mar 10 12:07:44] DEBUG[18599]: channel.c:1693 ast_hangup: Hanging up channel 'SIP/551-b770d980' [Mar 10 12:07:44] DEBUG[18599]: chan_sip.c:3310 sip_hangup: Hangup call SIP/551-b770d980, SIP callid 2919426183@192.168.1.97) [Mar 10 12:07:44] DEBUG[18599]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/551-b770d980 [Mar 10 12:07:44] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 551 [Mar 10 12:07:44] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 551 [Mar 10 12:07:44] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/551 - state 1 (Not in use) [Mar 10 12:07:44] DEBUG[18613]: app_queue.c:546 changethread: Device 'SIP/551' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Mar 10 12:07:45] NOTICE[18555]: chan_sip.c:7128 sip_reregister: -- Re-registration for 3173241052@sip.broadvoice.com [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4308 sip_alloc: Allocating new SIP dialog for 2c01f4fc3cc0317e799f35d0601622e1@24.123.23.170 - REGISTER (No RTP) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:7287 transmit_register: Scheduled a registration timeout for sip.broadvoice.com id #229 [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:7334 transmit_register: >>> Re-using Auth data for 3173241052@sip.broadvoice.com [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:sip.broadvoice.com SIP/2.0 (39) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK2b25ef3d;rport (64) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: From: ;tag=as443c0f9e (56) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: (39) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 2c01f4fc3cc0317e799f35d0601622e1@24.123.23.170 (55) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: CSeq: 106 REGISTER (18) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Max-Forwards: 70 (16) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Authorization: Digest username="3173241052", realm="BroadWorks", algorithm=MD5, uri="sip:sip.broadvoice.com", nonce="BroadWorksXez4b7xcwTti1x0kBW", response="b761bb69ba806078fc6d4e5895d3bc72", opaque="", qop=auth, cnonce="7e5bb9f8", nc=00000004 (244) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Expires: 120 (12) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Contact: (39) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Event: registration (19) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Content-Length: 0 (17) [Mar 10 12:07:45] REGISTER 13 headers, 0 lines [Mar 10 12:07:45] REGISTER attempt 1 to 3173241052@sip.broadvoice.com [Mar 10 12:07:45] 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=z9hG4bK2b25ef3d;rport From: ;tag=as443c0f9e To: Call-ID: 2c01f4fc3cc0317e799f35d0601622e1@24.123.23.170 CSeq: 106 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="3173241052", realm="BroadWorks", algorithm=MD5, uri="sip:sip.broadvoice.com", nonce="BroadWorksXez4b7xcwTti1x0kBW", response="b761bb69ba806078fc6d4e5895d3bc72", opaque="", qop=auth, cnonce="7e5bb9f8", nc=00000004 Expires: 120 Contact: Event: registration Content-Length: 0 --- [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #230 [Mar 10 12:07:45] <--- 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=z9hG4bK58e95129 From: To: Call-ID: 000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165 CSeq: 11596 REGISTER User-Agent: CSCO/7 Contact: Authorization: Digest username="512",realm="asterisk",uri="sip:24.123.23.170",response="7581c6e7de590798503036a2eb452892",nonce="71e86f1c",algorithm=MD5 Content-Length: 0 Expires: 60 <-------------> [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK58e95129 (58) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: From: (40) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: (38) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165 (58) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: CSeq: 11596 REGISTER (20) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: User-Agent: CSCO/7 (18) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Contact: (37) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Authorization: Digest username="512",realm="asterisk",uri="sip:24.123.23.170",response="7581c6e7de590798503036a2eb452892",nonce="71e86f1c",algorithm=MD5 (152) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Expires: 60 (11) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: (0) [Mar 10 12:07:45] --- (11 headers 0 lines) --- [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:07:45] Using latest REGISTER request as basis request [Mar 10 12:07:45] Sending to 192.168.1.165 : 5060 (no NAT) [Mar 10 12:07:45] <--- 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=z9hG4bK58e95129;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165 CSeq: 11596 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:45] <--- 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=z9hG4bK58e95129;received=192.168.1.165 From: To: ;tag=as601fb678 Call-ID: 000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165 CSeq: 11596 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Sat, 10 Mar 2007 17:07:45 GMT Content-Length: 0 <------------> [Mar 10 12:07:45] DEBUG[18555]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/512 [Mar 10 12:07:45] Scheduling destruction of SIP dialog '000ff78d-ebb20004-0988fd2a-11c3188b@192.168.1.165' in 32000 ms (Method: REGISTER) [Mar 10 12:07:45] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 512 [Mar 10 12:07:45] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 512 [Mar 10 12:07:45] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/512 - state 1 (Not in use) [Mar 10 12:07:45] DEBUG[18614]: 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. [Mar 10 12:07:45] <--- SIP read from 147.135.12.128:5060 ---> SIP/2.0 200 OK Call-ID: 2c01f4fc3cc0317e799f35d0601622e1@24.123.23.170 CSeq: 106 REGISTER From: ;tag=as443c0f9e To: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK2b25ef3d Contact: Expires: 30 Event: registration Content-Length: 0 <-------------> [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: SIP/2.0 200 OK (14) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Call-ID: 2c01f4fc3cc0317e799f35d0601622e1@24.123.23.170 (55) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 106 REGISTER (18) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: From: ;tag=as443c0f9e (56) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: To: (39) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK2b25ef3d (58) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: (39) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Expires: 30 (11) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Event: registration (19) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (20) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: (0) [Mar 10 12:07:45] --- (10 headers 0 lines) --- [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #230 [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '2c01f4fc3cc0317e799f35d0601622e1@24.123.23.170' of Request 106: Match Not Found [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:12025 handle_response_register: Registration successful [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:12028 handle_response_register: Cancelling timeout 229 [Mar 10 12:07:45] Scheduling destruction of SIP dialog '2c01f4fc3cc0317e799f35d0601622e1@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:07:45] NOTICE[18555]: chan_sip.c:12080 handle_response_register: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 23 s) [Mar 10 12:07:45] <--- 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=z9hG4bK1e7e12ee From: To: Call-ID: 000ff78d-ebb20007-4ad25281-15520cf9@192.168.1.165 CSeq: 11602 REGISTER User-Agent: CSCO/7 Contact: Authorization: Digest username="515",realm="asterisk",uri="sip:24.123.23.170",response="86b7acf8ecbd01f0f873cbf686b72f8b",nonce="52145cc9",algorithm=MD5 Content-Length: 0 Expires: 60 <-------------> [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK1e7e12ee (58) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: From: (40) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: (38) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 000ff78d-ebb20007-4ad25281-15520cf9@192.168.1.165 (58) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: CSeq: 11602 REGISTER (20) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: User-Agent: CSCO/7 (18) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Contact: (37) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Authorization: Digest username="515",realm="asterisk",uri="sip:24.123.23.170",response="86b7acf8ecbd01f0f873cbf686b72f8b",nonce="52145cc9",algorithm=MD5 (152) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Expires: 60 (11) [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: (0) [Mar 10 12:07:45] --- (11 headers 0 lines) --- [Mar 10 12:07:45] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:07:45] Using latest REGISTER request as basis request [Mar 10 12:07:45] Sending to 192.168.1.165 : 5060 (no NAT) [Mar 10 12:07:45] <--- 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=z9hG4bK1e7e12ee;received=192.168.1.165 From: To: Call-ID: 000ff78d-ebb20007-4ad25281-15520cf9@192.168.1.165 CSeq: 11602 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:45] <--- 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=z9hG4bK1e7e12ee;received=192.168.1.165 From: To: ;tag=as0befcdde Call-ID: 000ff78d-ebb20007-4ad25281-15520cf9@192.168.1.165 CSeq: 11602 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Sat, 10 Mar 2007 17:07:45 GMT Content-Length: 0 <------------> [Mar 10 12:07:45] DEBUG[18555]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/515 [Mar 10 12:07:45] Scheduling destruction of SIP dialog '000ff78d-ebb20007-4ad25281-15520cf9@192.168.1.165' in 32000 ms (Method: REGISTER) [Mar 10 12:07:45] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 515 [Mar 10 12:07:45] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 515 [Mar 10 12:07:45] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/515 - state 1 (Not in use) [Mar 10 12:07:45] DEBUG[18615]: 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. [Mar 10 12:07:45] <--- SIP read from 192.168.1.85:5060 ---> ACK sip:551@24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKf25626fcf1dff5c1e47e3b5ff18b6705f CSeq: 34085 ACK To: ;tag=as59744546 Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 From: 523 ;tag=fb7c7b32eac4985d4d8676d2196e3753 User-Agent: Uniden SIP Phone p2 Ver BS4.77 <-------------> [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: ACK sip:551@24.123.23.170 SIP/2.0 (33) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKf25626fcf1dff5c1e47e3b5ff18b6705f (82) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 34085 ACK (15) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: ;tag=as59744546 (42) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 (55) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: From: 523 ;tag=fb7c7b32eac4985d4d8676d2196e3753 (74) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: (0) [Mar 10 12:07:47] --- (7 headers 0 lines) --- [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received ACK (6) - Command in SIP ACK [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #223 [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170' of Response 34085: Match Not Found [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:5637 reqprep: Strict routing enforced for session 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 [Mar 10 12:07:47] set_destination: Parsing for address/port to send to [Mar 10 12:07:47] set_destination: set destination to 192.168.1.85, port 5060 [Mar 10 12:07:47] Reliably Transmitting (no NAT) to 192.168.1.85:5060: BYE sip:523@192.168.1.85:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK015ad08d From: ;tag=as59744546 To: 523 ;tag=fb7c7b32eac4985d4d8676d2196e3753 Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #237 [Mar 10 12:07:47] Scheduling destruction of SIP dialog '6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170' in 32000 ms (Method: ACK) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:1864 retrans_pkt: SIP TIMER: Rescheduling retransmission #216 (1) SIP/2.0 - 1 [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:1878 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #216)) [Mar 10 12:07:47] Retransmitting #1 (no NAT) to 192.168.1.85:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKec06c07c3079ca1c8c928ee81120eaffc;received=192.168.1.85 From: 523 ;tag=fb7c7b32eac4985d4d8676d2196e3753 To: ;tag=as59744546 Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 CSeq: 34085 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 18522 18524 IN IP4 192.168.1.97 s=session c=IN IP4 192.168.1.97 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 --- [Mar 10 12:07:47] DEBUG[18555]: sched.c:204 sched_settime: Request to schedule in the past?!?! [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:1864 retrans_pkt: SIP TIMER: Rescheduling retransmission #217 (1) INVITE - 5 [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:1878 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #217)) [Mar 10 12:07:47] Retransmitting #1 (NAT) to 192.168.1.97:5060: INVITE sip:551@192.168.1.97:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK4276259f;rport From: ;tag=as08598f69 To: "Display Name" ;tag=2534858470 Contact: Call-ID: 2919426183@192.168.1.97 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 18522 18524 IN IP4 24.123.23.170 s=session c=IN IP4 24.123.23.170 t=0 0 m=audio 12970 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 --- [Mar 10 12:07:47] DEBUG[18555]: sched.c:204 sched_settime: Request to schedule in the past?!?! [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:1864 retrans_pkt: SIP TIMER: Rescheduling retransmission #219 (1) SIP/2.0 - 1 [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:1878 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #219)) [Mar 10 12:07:47] Retransmitting #1 (no NAT) to 192.168.1.85:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKec06c07c3079ca1c8c928ee81120eaffc;received=192.168.1.85 From: 523 ;tag=fb7c7b32eac4985d4d8676d2196e3753 To: ;tag=as59744546 Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 CSeq: 34085 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 18522 18525 IN IP4 192.168.1.97 s=session c=IN IP4 192.168.1.97 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 --- [Mar 10 12:07:47] DEBUG[18555]: sched.c:204 sched_settime: Request to schedule in the past?!?! [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:1864 retrans_pkt: SIP TIMER: Rescheduling retransmission #216 (2) SIP/2.0 - 1 [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:1878 retrans_pkt: ** SIP timers: Rescheduling retransmission 3 to 2000 ms (t1 500 ms (Retrans id #216)) [Mar 10 12:07:47] Retransmitting #2 (no NAT) to 192.168.1.85:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKec06c07c3079ca1c8c928ee81120eaffc;received=192.168.1.85 From: 523 ;tag=fb7c7b32eac4985d4d8676d2196e3753 To: ;tag=as59744546 Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 CSeq: 34085 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 18522 18524 IN IP4 192.168.1.97 s=session c=IN IP4 192.168.1.97 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 --- [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:1864 retrans_pkt: SIP TIMER: Rescheduling retransmission #217 (2) INVITE - 5 [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:1878 retrans_pkt: ** SIP timers: Rescheduling retransmission 3 to 2000 ms (t1 500 ms (Retrans id #217)) [Mar 10 12:07:47] Retransmitting #2 (NAT) to 192.168.1.97:5060: INVITE sip:551@192.168.1.97:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK4276259f;rport From: ;tag=as08598f69 To: "Display Name" ;tag=2534858470 Contact: Call-ID: 2919426183@192.168.1.97 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 18522 18524 IN IP4 24.123.23.170 s=session c=IN IP4 24.123.23.170 t=0 0 m=audio 12970 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 --- [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:1864 retrans_pkt: SIP TIMER: Rescheduling retransmission #219 (2) SIP/2.0 - 1 [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:1878 retrans_pkt: ** SIP timers: Rescheduling retransmission 3 to 2000 ms (t1 500 ms (Retrans id #219)) [Mar 10 12:07:47] Retransmitting #2 (no NAT) to 192.168.1.85:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKec06c07c3079ca1c8c928ee81120eaffc;received=192.168.1.85 From: 523 ;tag=fb7c7b32eac4985d4d8676d2196e3753 To: ;tag=as59744546 Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 CSeq: 34085 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 18522 18525 IN IP4 192.168.1.97 s=session c=IN IP4 192.168.1.97 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 --- [Mar 10 12:07:47] <--- SIP read from 192.168.1.97:5060 ---> SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK4276259f;rport=5060 From: ;tag=as08598f69 To: "Display Name" ;tag=2534858470 Call-ID: 2919426183@192.168.1.97 CSeq: 103 INVITE Content-Length: 0 <-------------> [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: SIP/2.0 481 Call Leg/Transaction Does Not Exist (47) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK4276259f;rport=5060 (69) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: From: ;tag=as08598f69 (44) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: "Display Name" ;tag=2534858470 (57) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 2919426183@192.168.1.97 (32) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: CSeq: 103 INVITE (16) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Content-Length: 0 (17) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: (0) [Mar 10 12:07:47] --- (7 headers 0 lines) --- [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:2069 __sip_ack: Acked pending invite 103 [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #217 [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '2919426183@192.168.1.97' of Request 103: Match Not Found [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:11621 handle_response_invite: SIP response 481 to standard invite [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:11626 handle_response_invite: Got response on call that is already terminated: 2919426183@192.168.1.97 (ignoring) [Mar 10 12:07:47] Really destroying SIP dialog '2919426183@192.168.1.97' Method: BYE [Mar 10 12:07:47] <--- SIP read from 192.168.1.85:5060 ---> ACK sip:551@24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKf25626fcf1dff5c1e47e3b5ff18b6705f CSeq: 34085 ACK To: ;tag=as59744546 Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 From: 523 ;tag=fb7c7b32eac4985d4d8676d2196e3753 User-Agent: Uniden SIP Phone p2 Ver BS4.77 <-------------> [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: ACK sip:551@24.123.23.170 SIP/2.0 (33) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKf25626fcf1dff5c1e47e3b5ff18b6705f (82) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 34085 ACK (15) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: ;tag=as59744546 (42) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 (55) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: From: 523 ;tag=fb7c7b32eac4985d4d8676d2196e3753 (74) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: (0) [Mar 10 12:07:47] --- (7 headers 0 lines) --- [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received ACK (6) - Command in SIP ACK [Mar 10 12:07:47] <--- SIP read from 192.168.1.85:5060 ---> INVITE sip:593@24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKz9f560f234760d6220f9cb573a7e43127 Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 388021 INVITE From: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec To: ;tag=as628c2367 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: 269 v=0 o=- 1791739077 204408 IN IP4 192.168.1.85 s=- c=IN IP4 192.168.1.85 t=0 0 m=audio 30016 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15,16,17,18 a=sendrecv a=ptime:20 <-------------> [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: INVITE sip:593@24.123.23.170 SIP/2.0 (36) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKz9f560f234760d6220f9cb573a7e43127 (82) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 (55) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: CSeq: 388021 INVITE (19) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec (74) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: ;tag=as628c2367 (42) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: (36) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Content-Type: application/sdp (29) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Max-Forwards: 70 (16) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Content-Length: 269 (19) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: o=- 1791739077 204408 IN IP4 192.168.1.85 (41) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: s=- (3) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: c=IN IP4 192.168.1.85 (21) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: m=audio 30016 RTP/AVP 0 8 18 101 (32) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-15,16,17,18 (24) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=sendrecv (10) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:07:47] --- (13 headers 13 lines) --- [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Mar 10 12:07:47] Sending to 192.168.1.85 : 5060 (no NAT) [Mar 10 12:07:47] Found RTP audio format 0 [Mar 10 12:07:47] Found RTP audio format 8 [Mar 10 12:07:47] Found RTP audio format 18 [Mar 10 12:07:47] Found RTP audio format 101 [Mar 10 12:07:47] Peer audio RTP is at port 192.168.1.85:30016 [Mar 10 12:07:47] Found description format PCMU for ID 0 [Mar 10 12:07:47] Found description format PCMA for ID 8 [Mar 10 12:07:47] Found description format G729 for ID 18 [Mar 10 12:07:47] Found description format telephone-event for ID 101 [Mar 10 12:07:47] Got unsupported a:fmtp in SDP offer [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:5123 process_sdp: T38 state changed to 0 on channel SIP/523-087dd9a8 [Mar 10 12:07:47] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) [Mar 10 12:07:47] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 10 12:07:47] Peer audio RTP is at port 192.168.1.85:30016 [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:5203 process_sdp: We're settling with these formats: 0xc (ulaw|alaw) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:5210 process_sdp: We have an owner, now see if we need to change this call [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:13431 handle_request_invite: Got a SIP re-invite for call 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:13526 handle_request_invite: SIP/523-087dd9a8: This call is UP.... [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:6414 transmit_response_with_sdp: Setting framing from config on incoming call [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:6182 add_sdp: ** Our capability: 0xc (ulaw|alaw) Video flag: True [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:6183 add_sdp: ** Our prefcodec: 0x4 (ulaw) [Mar 10 12:07:47] Audio is at 24.123.23.170 port 10848 [Mar 10 12:07:47] Adding codec 0x4 (ulaw) to SDP [Mar 10 12:07:47] Adding codec 0x8 (alaw) to SDP [Mar 10 12:07:47] Adding non-codec 0x1 (telephone-event) to SDP [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:6314 add_sdp: -- Done with adding codecs to SDP [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:6359 add_sdp: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Mar 10 12:07:47] <--- Reliably Transmitting (no NAT) to 192.168.1.85:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKz9f560f234760d6220f9cb573a7e43127;received=192.168.1.85 From: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec To: ;tag=as628c2367 Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 388021 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 18522 18526 IN IP4 192.168.1.83 s=session c=IN IP4 192.168.1.83 t=0 0 m=audio 51000 RTP/AVP 0 8 101 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 <------------> [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #239 [Mar 10 12:07:47] DEBUG[18591]: chan_sip.c:16860 sip_set_rtp_peer: Sending reinvite on SIP 'a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83' - It's audio soon redirected to IP 192.168.1.85 [Mar 10 12:07:47] DEBUG[18591]: chan_sip.c:5637 reqprep: Strict routing enforced for session a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 [Mar 10 12:07:47] set_destination: Parsing for address/port to send to [Mar 10 12:07:47] set_destination: set destination to 192.168.1.83, port 5060 [Mar 10 12:07:47] DEBUG[18591]: chan_sip.c:6182 add_sdp: ** Our capability: 0xc (ulaw|alaw) Video flag: True [Mar 10 12:07:47] DEBUG[18591]: chan_sip.c:6183 add_sdp: ** Our prefcodec: 0x0 (nothing) [Mar 10 12:07:47] Audio is at 24.123.23.170 port 14996 [Mar 10 12:07:47] Adding codec 0x4 (ulaw) to SDP [Mar 10 12:07:47] Adding codec 0x8 (alaw) to SDP [Mar 10 12:07:47] Adding non-codec 0x1 (telephone-event) to SDP [Mar 10 12:07:47] DEBUG[18591]: chan_sip.c:6314 add_sdp: -- Done with adding codecs to SDP [Mar 10 12:07:47] DEBUG[18591]: chan_sip.c:6359 add_sdp: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Mar 10 12:07:47] DEBUG[18591]: chan_sip.c:1619 initialize_initreq: Initializing already initialized SIP dialog a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 (presumably reinvite) [Mar 10 12:07:47] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 0: INVITE sip:593@192.168.1.83:5060 SIP/2.0 (40) [Mar 10 12:07:47] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK3749480d (58) [Mar 10 12:07:47] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 2: From: 523 ;tag=as43feeb80 (53) [Mar 10 12:07:47] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 3: To: 593 ;tag=48204f4e1c15803 (56) [Mar 10 12:07:47] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 4: Contact: (32) [Mar 10 12:07:47] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 5: Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 (54) [Mar 10 12:07:47] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 6: CSeq: 104 INVITE (16) [Mar 10 12:07:47] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Mar 10 12:07:47] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:07:47] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Mar 10 12:07:47] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 10: Supported: replaces (19) [Mar 10 12:07:47] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Mar 10 12:07:47] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 12: Content-Type: application/sdp (29) [Mar 10 12:07:47] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 13: Content-Length: 264 (19) [Mar 10 12:07:47] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 14: (0) [Mar 10 12:07:47] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:07:47] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: o=root 18522 18525 IN IP4 192.168.1.85 (38) [Mar 10 12:07:47] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: s=session (9) [Mar 10 12:07:47] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: c=IN IP4 192.168.1.85 (21) [Mar 10 12:07:47] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:07:47] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: m=audio 30016 RTP/AVP 0 8 101 (29) [Mar 10 12:07:47] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:07:47] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Mar 10 12:07:47] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:07:47] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-16 (15) [Mar 10 12:07:47] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=silenceSupp:off - - - - (25) [Mar 10 12:07:47] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:07:47] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=sendrecv (10) [Mar 10 12:07:47] Reliably Transmitting (no NAT) to 192.168.1.83:5060: INVITE sip:593@192.168.1.83:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK3749480d From: 523 ;tag=as43feeb80 To: 593 ;tag=48204f4e1c15803 Contact: Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 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: 264 v=0 o=root 18522 18525 IN IP4 192.168.1.85 s=session c=IN IP4 192.168.1.85 t=0 0 m=audio 30016 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 --- [Mar 10 12:07:47] DEBUG[18591]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #240 [Mar 10 12:07:47] DEBUG[18591]: channel.c:2845 set_format: Set channel SIP/593-b77052f8 to write format ulaw [Mar 10 12:07:47] -- Stopped music on hold on SIP/593-b77052f8 [Mar 10 12:07:47] DEBUG[18591]: channel.c:1997 ast_settimeout: Scheduling timer at 0 sample intervals [Mar 10 12:07:47] DEBUG[18591]: rtp.c:2806 bridge_native_loop: Oooh, 'SIP/523-087dd9a8' changed end address to 192.168.1.85:30016 (format 268) [Mar 10 12:07:47] DEBUG[18591]: rtp.c:2808 bridge_native_loop: Oooh, 'SIP/523-087dd9a8' changed end vaddress to 0.0.0.0:0 (format 268) [Mar 10 12:07:47] DEBUG[18591]: rtp.c:2810 bridge_native_loop: Oooh, 'SIP/523-087dd9a8' was 0.0.0.0:0/(format 4) [Mar 10 12:07:47] DEBUG[18591]: rtp.c:2812 bridge_native_loop: Oooh, 'SIP/523-087dd9a8' was 0.0.0.0:0/(format 4) [Mar 10 12:07:47] DEBUG[18591]: chan_sip.c:16865 sip_set_rtp_peer: Deferring reinvite on SIP 'a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83' - It's audio will be redirected to IP 192.168.1.85 [Mar 10 12:07:47] DEBUG[18591]: rtp.c:2670 ast_rtp_write: Ooh, format changed from unknown to ulaw [Mar 10 12:07:47] DEBUG[18591]: rtp.c:2687 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Mar 10 12:07:47] <--- SIP read from 192.168.1.85:5060 ---> INVITE sip:593@24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKz9f560f234760d6220f9cb573a7e43127 Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 388021 INVITE From: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec To: ;tag=as628c2367 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: 269 v=0 o=- 1791739077 204408 IN IP4 192.168.1.85 s=- c=IN IP4 192.168.1.85 t=0 0 m=audio 30016 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15,16,17,18 a=sendrecv a=ptime:20 <-------------> [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: INVITE sip:593@24.123.23.170 SIP/2.0 (36) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKz9f560f234760d6220f9cb573a7e43127 (82) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 (55) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: CSeq: 388021 INVITE (19) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec (74) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: ;tag=as628c2367 (42) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: (36) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Content-Type: application/sdp (29) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Max-Forwards: 70 (16) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Content-Length: 269 (19) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: o=- 1791739077 204408 IN IP4 192.168.1.85 (41) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: s=- (3) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: c=IN IP4 192.168.1.85 (21) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: m=audio 30016 RTP/AVP 0 8 18 101 (32) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-15,16,17,18 (24) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=sendrecv (10) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:07:47] --- (13 headers 13 lines) --- [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (INVITE Seqno 388021, ours 388021) [Mar 10 12:07:47] Ignoring this INVITE request [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:13433 handle_request_invite: Got a SIP re-transmit of INVITE for call 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:13526 handle_request_invite: SIP/523-087dd9a8: This call is UP.... [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:6414 transmit_response_with_sdp: Setting framing from config on incoming call [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:6182 add_sdp: ** Our capability: 0xc (ulaw|alaw) Video flag: True [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:6183 add_sdp: ** Our prefcodec: 0x4 (ulaw) [Mar 10 12:07:47] Audio is at 24.123.23.170 port 10848 [Mar 10 12:07:47] Adding codec 0x4 (ulaw) to SDP [Mar 10 12:07:47] Adding codec 0x8 (alaw) to SDP [Mar 10 12:07:47] Adding non-codec 0x1 (telephone-event) to SDP [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:6314 add_sdp: -- Done with adding codecs to SDP [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:6359 add_sdp: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Mar 10 12:07:47] <--- Reliably Transmitting (no NAT) to 192.168.1.85:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKz9f560f234760d6220f9cb573a7e43127;received=192.168.1.85 From: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec To: ;tag=as628c2367 Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 388021 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 18522 18527 IN IP4 192.168.1.83 s=session c=IN IP4 192.168.1.83 t=0 0 m=audio 51000 RTP/AVP 0 8 101 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 <------------> [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #242 [Mar 10 12:07:47] DEBUG[18591]: rtp.c:2687 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Mar 10 12:07:47] <--- SIP read from 192.168.1.85:5060 ---> SIP/2.0 200 OK To: 523 ;tag=fb7c7b32eac4985d4d8676d2196e3753 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK015ad08d From: ;tag=as59744546 Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@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 <-------------> [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: SIP/2.0 200 OK (14) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: To: 523 ;tag=fb7c7b32eac4985d4d8676d2196e3753 (72) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK015ad08d (58) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: From: ;tag=as59744546 (44) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 (55) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: CSeq: 104 BYE (13) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Content-Length: 0 (17) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: (0) [Mar 10 12:07:47] --- (9 headers 0 lines) --- [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #237 [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170' of Request 104: Match Not Found [Mar 10 12:07:47] SIP Response message for INCOMING dialog BYE arrived [Mar 10 12:07:47] <--- SIP read from 192.168.1.83:5060 ---> SIP/2.0 200 OK Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 CSeq: 104 INVITE From: 523 ;tag=as43feeb80 To: 593 ;tag=48204f4e1c15803 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK3749480d Content-Length: 266 Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE Content-Type: application/sdp Supported: replaces Contact: 593 User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 v=0 o=MxSIP 0 1143068287 IN IP4 192.168.1.83 s=SIP Call c=IN IP4 192.168.1.83 t=0 0 m=audio 51000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:on - - - - <-------------> [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: SIP/2.0 200 OK (14) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 (54) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 104 INVITE (16) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: From: 523 ;tag=as43feeb80 (53) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: To: 593 ;tag=48204f4e1c15803 (56) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK3749480d (58) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Content-Length: 266 (19) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE (55) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Content-Type: application/sdp (29) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Supported: replaces (19) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Contact: 593 (40) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 (42) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: (0) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: o=MxSIP 0 1143068287 IN IP4 192.168.1.83 (40) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: s=SIP Call (10) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: c=IN IP4 192.168.1.83 (21) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: m=audio 51000 RTP/AVP 0 8 101 (29) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-15 (15) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=sendrecv (10) [Mar 10 12:07:47] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=silenceSupp:on - - - - (24) [Mar 10 12:07:48] DEBUG[18545]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel IAX2/614 [Mar 10 12:07:49] --- (12 headers 13 lines) --- [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:2069 __sip_ack: Acked pending invite 104 [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #240 [Mar 10 12:07:49] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for IAX2 - 614 [Mar 10 12:07:49] DEBUG[18527]: chan_iax2.c:9636 iax2_devicestate: Checking device state for device 614 [Mar 10 12:07:49] DEBUG[18527]: chan_iax2.c:9644 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 614? addr=-1493063488, defaddr=0 maxms=0, lastms=0 [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on 'a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83' of Request 104: Match Not Found [Mar 10 12:07:49] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for IAX2/614 - state 1 (Not in use) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:11619 handle_response_invite: SIP response 200 to RE-invite on outgoing call a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 [Mar 10 12:07:49] DEBUG[18616]: 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. [Mar 10 12:07:49] Found RTP audio format 0 [Mar 10 12:07:49] Found RTP audio format 8 [Mar 10 12:07:49] Found RTP audio format 101 [Mar 10 12:07:49] Peer audio RTP is at port 192.168.1.83:51000 [Mar 10 12:07:49] Found description format PCMU for ID 0 [Mar 10 12:07:49] Found description format PCMA for ID 8 [Mar 10 12:07:49] Found description format telephone-event for ID 101 [Mar 10 12:07:49] Got unsupported a:fmtp in SDP offer [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:5123 process_sdp: T38 state changed to 0 on channel SIP/593-b77052f8 [Mar 10 12:07:49] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) [Mar 10 12:07:49] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 10 12:07:49] Peer audio RTP is at port 192.168.1.83:51000 [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:5203 process_sdp: We're settling with these formats: 0xc (ulaw|alaw) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:5210 process_sdp: We have an owner, now see if we need to change this call [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:3001 update_call_counter: Updating call counter for incoming call [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:7903 build_route: build_route: Retaining previous route: [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:11752 handle_response_invite: Strange... The other side of the bridge does not have a udptl struct [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:11757 handle_response_invite: T38 state changed to 0 on channel SIP [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:11760 handle_response_invite: T38 state changed to 0 on channel SIP/593-b77052f8 [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:5637 reqprep: Strict routing enforced for session a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 [Mar 10 12:07:49] set_destination: Parsing for address/port to send to [Mar 10 12:07:49] set_destination: set destination to 192.168.1.83, port 5060 [Mar 10 12:07:49] Transmitting (no NAT) to 192.168.1.83:5060: ACK sip:593@192.168.1.83:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK7e5b8869 From: 523 ;tag=as43feeb80 To: 593 ;tag=48204f4e1c15803 Contact: Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 CSeq: 104 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:11602 check_pendings: Sending pending reinvite on 'a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83' [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:5637 reqprep: Strict routing enforced for session a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 [Mar 10 12:07:49] set_destination: Parsing for address/port to send to [Mar 10 12:07:49] set_destination: set destination to 192.168.1.83, port 5060 [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:6182 add_sdp: ** Our capability: 0xc (ulaw|alaw) Video flag: True [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:6183 add_sdp: ** Our prefcodec: 0x0 (nothing) [Mar 10 12:07:49] Audio is at 24.123.23.170 port 14996 [Mar 10 12:07:49] Adding codec 0x4 (ulaw) to SDP [Mar 10 12:07:49] Adding codec 0x8 (alaw) to SDP [Mar 10 12:07:49] Adding non-codec 0x1 (telephone-event) to SDP [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:6314 add_sdp: -- Done with adding codecs to SDP [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:6359 add_sdp: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:1619 initialize_initreq: Initializing already initialized SIP dialog a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 (presumably reinvite) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: INVITE sip:593@192.168.1.83:5060 SIP/2.0 (40) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK0140090b (58) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: From: 523 ;tag=as43feeb80 (53) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: 593 ;tag=48204f4e1c15803 (56) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Contact: (32) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 (54) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: CSeq: 105 INVITE (16) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: replaces (19) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Content-Type: application/sdp (29) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: Content-Length: 264 (19) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 14: (0) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: o=root 18522 18526 IN IP4 192.168.1.85 (38) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: s=session (9) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: c=IN IP4 192.168.1.85 (21) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: m=audio 30016 RTP/AVP 0 8 101 (29) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-16 (15) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=silenceSupp:off - - - - (25) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=sendrecv (10) [Mar 10 12:07:49] Reliably Transmitting (no NAT) to 192.168.1.83:5060: INVITE sip:593@192.168.1.83:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK0140090b From: 523 ;tag=as43feeb80 To: 593 ;tag=48204f4e1c15803 Contact: Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 CSeq: 105 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: 264 v=0 o=root 18522 18526 IN IP4 192.168.1.85 s=session c=IN IP4 192.168.1.85 t=0 0 m=audio 30016 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 --- [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #243 [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:1864 retrans_pkt: SIP TIMER: Rescheduling retransmission #239 (1) SIP/2.0 - 1 [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:1878 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #239)) [Mar 10 12:07:49] Retransmitting #1 (no NAT) to 192.168.1.85:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKz9f560f234760d6220f9cb573a7e43127;received=192.168.1.85 From: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec To: ;tag=as628c2367 Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 388021 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 18522 18526 IN IP4 192.168.1.83 s=session c=IN IP4 192.168.1.83 t=0 0 m=audio 51000 RTP/AVP 0 8 101 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 --- [Mar 10 12:07:49] DEBUG[18555]: sched.c:204 sched_settime: Request to schedule in the past?!?! [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:1864 retrans_pkt: SIP TIMER: Rescheduling retransmission #242 (1) SIP/2.0 - 1 [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:1878 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #242)) [Mar 10 12:07:49] Retransmitting #1 (no NAT) to 192.168.1.85:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKz9f560f234760d6220f9cb573a7e43127;received=192.168.1.85 From: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec To: ;tag=as628c2367 Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 388021 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 18522 18527 IN IP4 192.168.1.83 s=session c=IN IP4 192.168.1.83 t=0 0 m=audio 51000 RTP/AVP 0 8 101 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 --- [Mar 10 12:07:49] DEBUG[18555]: sched.c:204 sched_settime: Request to schedule in the past?!?! [Mar 10 12:07:49] DEBUG[18555]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/525 [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:1864 retrans_pkt: SIP TIMER: Rescheduling retransmission #216 (3) SIP/2.0 - 1 [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:1878 retrans_pkt: ** SIP timers: Rescheduling retransmission 4 to 4000 ms (t1 500 ms (Retrans id #216)) [Mar 10 12:07:49] Retransmitting #3 (no NAT) to 192.168.1.85:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKec06c07c3079ca1c8c928ee81120eaffc;received=192.168.1.85 From: 523 ;tag=fb7c7b32eac4985d4d8676d2196e3753 To: ;tag=as59744546 Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 CSeq: 34085 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 18522 18524 IN IP4 192.168.1.97 s=session c=IN IP4 192.168.1.97 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 --- [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:1864 retrans_pkt: SIP TIMER: Rescheduling retransmission #219 (3) SIP/2.0 - 1 [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:1878 retrans_pkt: ** SIP timers: Rescheduling retransmission 4 to 4000 ms (t1 500 ms (Retrans id #219)) [Mar 10 12:07:49] Retransmitting #3 (no NAT) to 192.168.1.85:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKec06c07c3079ca1c8c928ee81120eaffc;received=192.168.1.85 From: 523 ;tag=fb7c7b32eac4985d4d8676d2196e3753 To: ;tag=as59744546 Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 CSeq: 34085 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 18522 18525 IN IP4 192.168.1.97 s=session c=IN IP4 192.168.1.97 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 --- [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:1864 retrans_pkt: SIP TIMER: Rescheduling retransmission #239 (2) SIP/2.0 - 1 [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:1878 retrans_pkt: ** SIP timers: Rescheduling retransmission 3 to 2000 ms (t1 500 ms (Retrans id #239)) [Mar 10 12:07:49] Retransmitting #2 (no NAT) to 192.168.1.85:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKz9f560f234760d6220f9cb573a7e43127;received=192.168.1.85 From: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec To: ;tag=as628c2367 Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 388021 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 18522 18526 IN IP4 192.168.1.83 s=session c=IN IP4 192.168.1.83 t=0 0 m=audio 51000 RTP/AVP 0 8 101 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 --- [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:1864 retrans_pkt: SIP TIMER: Rescheduling retransmission #242 (2) SIP/2.0 - 1 [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:1878 retrans_pkt: ** SIP timers: Rescheduling retransmission 3 to 2000 ms (t1 500 ms (Retrans id #242)) [Mar 10 12:07:49] Retransmitting #2 (no NAT) to 192.168.1.85:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKz9f560f234760d6220f9cb573a7e43127;received=192.168.1.85 From: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec To: ;tag=as628c2367 Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 388021 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 18522 18527 IN IP4 192.168.1.83 s=session c=IN IP4 192.168.1.83 t=0 0 m=audio 51000 RTP/AVP 0 8 101 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 --- [Mar 10 12:07:49] <--- SIP read from 192.168.1.97:5060 ---> SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK4276259f;rport=5060 From: ;tag=as08598f69 To: "Display Name" ;tag=2534858470 Call-ID: 2919426183@192.168.1.97 CSeq: 103 INVITE Content-Length: 0 <-------------> [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: SIP/2.0 481 Call Leg/Transaction Does Not Exist (47) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK4276259f;rport=5060 (69) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: From: ;tag=as08598f69 (44) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: "Display Name" ;tag=2534858470 (57) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 2919426183@192.168.1.97 (32) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: CSeq: 103 INVITE (16) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Content-Length: 0 (17) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: (0) [Mar 10 12:07:49] --- (7 headers 0 lines) --- [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:14774 sipsock_read: Invalid SIP message - rejected , no callid, len 298 [Mar 10 12:07:49] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 525 [Mar 10 12:07:49] <--- SIP read from 192.168.1.97:5060 ---> SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK4276259f;rport=5060 From: ;tag=as08598f69 To: "Display Name" ;tag=2534858470 Call-ID: 2919426183@192.168.1.97 CSeq: 103 INVITE Content-Length: 0 <-------------> [Mar 10 12:07:49] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 525 [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: SIP/2.0 481 Call Leg/Transaction Does Not Exist (47) [Mar 10 12:07:49] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/525 - state 5 (Unavailable) [Mar 10 12:07:49] DEBUG[18617]: app_queue.c:546 changethread: Device 'SIP/525' changed to state '5' (Unavailable) but we don't care because they're not a member of any queue. [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK4276259f;rport=5060 (69) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: From: ;tag=as08598f69 (44) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: "Display Name" ;tag=2534858470 (57) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 2919426183@192.168.1.97 (32) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: CSeq: 103 INVITE (16) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Content-Length: 0 (17) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: (0) [Mar 10 12:07:49] --- (7 headers 0 lines) --- [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:14774 sipsock_read: Invalid SIP message - rejected , no callid, len 298 [Mar 10 12:07:49] <--- SIP read from 192.168.1.97:5060 ---> <-------------> [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: (0) [Mar 10 12:07:49] --- (0 headers 0 lines) Nat keepalive --- [Mar 10 12:07:49] <--- SIP read from 192.168.1.85:5060 ---> ACK sip:593@24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKq7a080ca15e06daffaee77dc5a87b8385 CSeq: 388021 ACK To: ;tag=as628c2367 Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 From: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec User-Agent: Uniden SIP Phone p2 Ver BS4.77 <-------------> [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: ACK sip:593@24.123.23.170 SIP/2.0 (33) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKq7a080ca15e06daffaee77dc5a87b8385 (82) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 388021 ACK (16) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: ;tag=as628c2367 (42) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 (55) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: From: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec (74) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: (0) [Mar 10 12:07:49] --- (7 headers 0 lines) --- [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received ACK (6) - Command in SIP ACK [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #242 [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '53c067b5491f609931bc7a0965b73fd3@24.123.23.170' of Response 388021: Match Not Found [Mar 10 12:07:49] <--- SIP read from 192.168.1.83:5060 ---> SIP/2.0 200 OK Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 CSeq: 104 INVITE From: 523 ;tag=as43feeb80 To: 593 ;tag=48204f4e1c15803 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK3749480d Content-Length: 266 Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE Content-Type: application/sdp Supported: replaces Contact: 593 User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 v=0 o=MxSIP 0 1143068287 IN IP4 192.168.1.83 s=SIP Call c=IN IP4 192.168.1.83 t=0 0 m=audio 51000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:on - - - - <-------------> [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: SIP/2.0 200 OK (14) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 (54) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 104 INVITE (16) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: From: 523 ;tag=as43feeb80 (53) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: To: 593 ;tag=48204f4e1c15803 (56) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK3749480d (58) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Content-Length: 266 (19) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE (55) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Content-Type: application/sdp (29) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Supported: replaces (19) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Contact: 593 (40) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 (42) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: (0) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: o=MxSIP 0 1143068287 IN IP4 192.168.1.83 (40) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: s=SIP Call (10) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: c=IN IP4 192.168.1.83 (21) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: m=audio 51000 RTP/AVP 0 8 101 (29) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-15 (15) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=sendrecv (10) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=silenceSupp:on - - - - (24) [Mar 10 12:07:49] --- (12 headers 13 lines) --- [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on 'a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83' of Request 104: Match Found [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:11619 handle_response_invite: SIP response 200 to RE-invite on outgoing call a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 [Mar 10 12:07:49] Found RTP audio format 0 [Mar 10 12:07:49] Found RTP audio format 8 [Mar 10 12:07:49] Found RTP audio format 101 [Mar 10 12:07:49] Peer audio RTP is at port 192.168.1.83:51000 [Mar 10 12:07:49] Found description format PCMU for ID 0 [Mar 10 12:07:49] Found description format PCMA for ID 8 [Mar 10 12:07:49] Found description format telephone-event for ID 101 [Mar 10 12:07:49] Got unsupported a:fmtp in SDP offer [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:5123 process_sdp: T38 state changed to 0 on channel SIP/593-b77052f8 [Mar 10 12:07:49] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) [Mar 10 12:07:49] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 10 12:07:49] Peer audio RTP is at port 192.168.1.83:51000 [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:5203 process_sdp: We're settling with these formats: 0xc (ulaw|alaw) [Mar 10 12:07:49] DEBUG[18555]: chan_sip.c:5210 process_sdp: We have an owner, now see if we need to change this call [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:3001 update_call_counter: Updating call counter for incoming call [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:7903 build_route: build_route: Retaining previous route: [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:11752 handle_response_invite: Strange... The other side of the bridge does not have a udptl struct [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:11757 handle_response_invite: T38 state changed to 0 on channel SIP [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:11760 handle_response_invite: T38 state changed to 0 on channel SIP/593-b77052f8 [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:5637 reqprep: Strict routing enforced for session a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 [Mar 10 12:07:52] set_destination: Parsing for address/port to send to [Mar 10 12:07:52] set_destination: set destination to 192.168.1.83, port 5060 [Mar 10 12:07:52] Transmitting (no NAT) to 192.168.1.83:5060: ACK sip:593@192.168.1.83:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK136646bb From: 523 ;tag=as43feeb80 To: 593 ;tag=48204f4e1c15803 Contact: Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 CSeq: 104 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:1864 retrans_pkt: SIP TIMER: Rescheduling retransmission #243 (1) INVITE - 5 [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:1878 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #243)) [Mar 10 12:07:52] Retransmitting #1 (no NAT) to 192.168.1.83:5060: INVITE sip:593@192.168.1.83:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK0140090b From: 523 ;tag=as43feeb80 To: 593 ;tag=48204f4e1c15803 Contact: Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 CSeq: 105 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: 264 v=0 o=root 18522 18526 IN IP4 192.168.1.85 s=session c=IN IP4 192.168.1.85 t=0 0 m=audio 30016 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 --- [Mar 10 12:07:52] DEBUG[18555]: sched.c:204 sched_settime: Request to schedule in the past?!?! [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:1864 retrans_pkt: SIP TIMER: Rescheduling retransmission #239 (3) SIP/2.0 - 1 [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:1878 retrans_pkt: ** SIP timers: Rescheduling retransmission 4 to 4000 ms (t1 500 ms (Retrans id #239)) [Mar 10 12:07:52] Retransmitting #3 (no NAT) to 192.168.1.85:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKz9f560f234760d6220f9cb573a7e43127;received=192.168.1.85 From: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec To: ;tag=as628c2367 Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 388021 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 18522 18526 IN IP4 192.168.1.83 s=session c=IN IP4 192.168.1.83 t=0 0 m=audio 51000 RTP/AVP 0 8 101 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 --- [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:1864 retrans_pkt: SIP TIMER: Rescheduling retransmission #243 (2) INVITE - 5 [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:1878 retrans_pkt: ** SIP timers: Rescheduling retransmission 3 to 2000 ms (t1 500 ms (Retrans id #243)) [Mar 10 12:07:52] Retransmitting #2 (no NAT) to 192.168.1.83:5060: INVITE sip:593@192.168.1.83:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK0140090b From: 523 ;tag=as43feeb80 To: 593 ;tag=48204f4e1c15803 Contact: Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 CSeq: 105 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: 264 v=0 o=root 18522 18526 IN IP4 192.168.1.85 s=session c=IN IP4 192.168.1.85 t=0 0 m=audio 30016 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 --- [Mar 10 12:07:49] DEBUG[18544]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel IAX2/651 [Mar 10 12:07:52] <--- 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=z9hG4bKf098d5229453e8cd803d7b2ac98a0f699 Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 48250 REGISTER From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 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="1435817e", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="b9f5b2310a13162083b4809817c40870" <-------------> [Mar 10 12:07:52] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for IAX2 - 651 [Mar 10 12:07:52] DEBUG[18527]: chan_iax2.c:9636 iax2_devicestate: Checking device state for device 651 [Mar 10 12:07:52] DEBUG[18527]: chan_iax2.c:9644 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 651? addr=-1459509056, defaddr=0 maxms=0, lastms=0 [Mar 10 12:07:52] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for IAX2/651 - state 1 (Not in use) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKf098d5229453e8cd803d7b2ac98a0f699 (82) [Mar 10 12:07:52] DEBUG[18546]: chan_iax2.c:6538 socket_process: Packet arrived out of order (expecting 1, got 0) (frametype = 6, subclass = 13) [Mar 10 12:07:52] DEBUG[18546]: chan_iax2.c:6545 socket_process: Acking anyway [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Call-ID: 55ae66469f7f@24.123.23.170 (35) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: CSeq: 48250 REGISTER (20) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 (66) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="1435817e", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="b9f5b2310a13162083b4809817c40870" (167) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:07:52] --- (13 headers 0 lines) --- [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4308 sip_alloc: Allocating new SIP dialog for 55ae66469f7f@24.123.23.170 - REGISTER (No RTP) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:07:52] Using latest REGISTER request as basis request [Mar 10 12:07:52] Sending to 192.168.1.93 : 5060 (no NAT) [Mar 10 12:07:52] <--- 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=z9hG4bKf098d5229453e8cd803d7b2ac98a0f699;received=192.168.1.93 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 To: Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 48250 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:52] <--- 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=z9hG4bKf098d5229453e8cd803d7b2ac98a0f699;received=192.168.1.93 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 To: ;tag=as549fb064 Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 48250 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3ec2039b" Content-Length: 0 <------------> [Mar 10 12:07:52] Scheduling destruction of SIP dialog '55ae66469f7f@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:07:52] <--- 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=z9hG4bKk52a176f208afcf982ef28c96861d4306 Call-ID: 55ae664696c0@24.123.23.170 CSeq: 43041 REGISTER From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 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="1a957140", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="fbea8c836c27273ae975e32edd59f358" <-------------> [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.98:5060;branch=z9hG4bKk52a176f208afcf982ef28c96861d4306 (82) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Call-ID: 55ae664696c0@24.123.23.170 (35) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: CSeq: 43041 REGISTER (20) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 (66) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="1a957140", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="fbea8c836c27273ae975e32edd59f358" (167) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:07:52] --- (13 headers 0 lines) --- [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4308 sip_alloc: Allocating new SIP dialog for 55ae664696c0@24.123.23.170 - REGISTER (No RTP) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:07:52] Using latest REGISTER request as basis request [Mar 10 12:07:52] Sending to 192.168.1.98 : 5060 (no NAT) [Mar 10 12:07:52] <--- 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=z9hG4bKk52a176f208afcf982ef28c96861d4306;received=192.168.1.98 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 To: Call-ID: 55ae664696c0@24.123.23.170 CSeq: 43041 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:52] <--- 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=z9hG4bKk52a176f208afcf982ef28c96861d4306;received=192.168.1.98 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 To: ;tag=as08f52ac6 Call-ID: 55ae664696c0@24.123.23.170 CSeq: 43041 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2e41eec5" Content-Length: 0 <------------> [Mar 10 12:07:52] Scheduling destruction of SIP dialog '55ae664696c0@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:07:52] <--- SIP read from 192.168.1.83:5060 ---> SIP/2.0 200 OK Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 CSeq: 104 INVITE From: 523 ;tag=as43feeb80 To: 593 ;tag=48204f4e1c15803 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK3749480d Content-Length: 266 Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE Content-Type: application/sdp Supported: replaces Contact: 593 User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 v=0 o=MxSIP 0 1143068287 IN IP4 192.168.1.83 s=SIP Call c=IN IP4 192.168.1.83 t=0 0 m=audio 51000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:on - - - - <-------------> [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: SIP/2.0 200 OK (14) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 (54) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 104 INVITE (16) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: From: 523 ;tag=as43feeb80 (53) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: To: 593 ;tag=48204f4e1c15803 (56) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK3749480d (58) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Content-Length: 266 (19) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE (55) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Content-Type: application/sdp (29) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Supported: replaces (19) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Contact: 593 (40) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 (42) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: (0) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: o=MxSIP 0 1143068287 IN IP4 192.168.1.83 (40) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: s=SIP Call (10) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: c=IN IP4 192.168.1.83 (21) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: m=audio 51000 RTP/AVP 0 8 101 (29) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-15 (15) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=sendrecv (10) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=silenceSupp:on - - - - (24) [Mar 10 12:07:52] --- (12 headers 13 lines) --- [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on 'a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83' of Request 104: Match Found [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:11619 handle_response_invite: SIP response 200 to RE-invite on outgoing call a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 [Mar 10 12:07:52] Found RTP audio format 0 [Mar 10 12:07:52] Found RTP audio format 8 [Mar 10 12:07:52] Found RTP audio format 101 [Mar 10 12:07:52] Peer audio RTP is at port 192.168.1.83:51000 [Mar 10 12:07:52] Found description format PCMU for ID 0 [Mar 10 12:07:52] Found description format PCMA for ID 8 [Mar 10 12:07:52] Found description format telephone-event for ID 101 [Mar 10 12:07:52] Got unsupported a:fmtp in SDP offer [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:5123 process_sdp: T38 state changed to 0 on channel SIP/593-b77052f8 [Mar 10 12:07:52] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) [Mar 10 12:07:52] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 10 12:07:52] Peer audio RTP is at port 192.168.1.83:51000 [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:5203 process_sdp: We're settling with these formats: 0xc (ulaw|alaw) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:5210 process_sdp: We have an owner, now see if we need to change this call [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:3001 update_call_counter: Updating call counter for incoming call [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:7903 build_route: build_route: Retaining previous route: [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:11752 handle_response_invite: Strange... The other side of the bridge does not have a udptl struct [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:11757 handle_response_invite: T38 state changed to 0 on channel SIP [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:11760 handle_response_invite: T38 state changed to 0 on channel SIP/593-b77052f8 [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:5637 reqprep: Strict routing enforced for session a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 [Mar 10 12:07:52] set_destination: Parsing for address/port to send to [Mar 10 12:07:52] set_destination: set destination to 192.168.1.83, port 5060 [Mar 10 12:07:52] Transmitting (no NAT) to 192.168.1.83:5060: ACK sip:593@192.168.1.83:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK3524c60e From: 523 ;tag=as43feeb80 To: 593 ;tag=48204f4e1c15803 Contact: Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 CSeq: 104 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 10 12:07:52] DEBUG[18618]: 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. [Mar 10 12:07:52] <--- 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=z9hG4bKf098d5229453e8cd803d7b2ac98a0f699 Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 48250 REGISTER From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 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="1435817e", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="b9f5b2310a13162083b4809817c40870" <-------------> [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKf098d5229453e8cd803d7b2ac98a0f699 (82) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Call-ID: 55ae66469f7f@24.123.23.170 (35) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: CSeq: 48250 REGISTER (20) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 (66) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="1435817e", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="b9f5b2310a13162083b4809817c40870" (167) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:07:52] --- (13 headers 0 lines) --- [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 48250, ours 48250) [Mar 10 12:07:52] Using latest REGISTER request as basis request [Mar 10 12:07:52] Sending to 192.168.1.93 : 5060 (no NAT) [Mar 10 12:07:52] <--- 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=z9hG4bKf098d5229453e8cd803d7b2ac98a0f699;received=192.168.1.93 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 To: Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 48250 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:52] NOTICE[18555]: chan_sip.c:8135 check_auth: Correct auth, but based on stale nonce received from '' [Mar 10 12:07:52] <--- 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=z9hG4bKf098d5229453e8cd803d7b2ac98a0f699;received=192.168.1.93 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 To: ;tag=as549fb064 Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 48250 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="530ec8aa", stale=true Content-Length: 0 <------------> [Mar 10 12:07:52] Scheduling destruction of SIP dialog '55ae66469f7f@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:07:52] <--- 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=z9hG4bKk52a176f208afcf982ef28c96861d4306 Call-ID: 55ae664696c0@24.123.23.170 CSeq: 43041 REGISTER From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 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="1a957140", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="fbea8c836c27273ae975e32edd59f358" <-------------> [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.98:5060;branch=z9hG4bKk52a176f208afcf982ef28c96861d4306 (82) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Call-ID: 55ae664696c0@24.123.23.170 (35) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: CSeq: 43041 REGISTER (20) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 (66) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="1a957140", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="fbea8c836c27273ae975e32edd59f358" (167) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:07:52] --- (13 headers 0 lines) --- [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 43041, ours 43041) [Mar 10 12:07:52] Using latest REGISTER request as basis request [Mar 10 12:07:52] Sending to 192.168.1.98 : 5060 (no NAT) [Mar 10 12:07:52] <--- 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=z9hG4bKk52a176f208afcf982ef28c96861d4306;received=192.168.1.98 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 To: Call-ID: 55ae664696c0@24.123.23.170 CSeq: 43041 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:52] NOTICE[18555]: chan_sip.c:8135 check_auth: Correct auth, but based on stale nonce received from '' [Mar 10 12:07:52] <--- 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=z9hG4bKk52a176f208afcf982ef28c96861d4306;received=192.168.1.98 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 To: ;tag=as08f52ac6 Call-ID: 55ae664696c0@24.123.23.170 CSeq: 43041 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6dc6d41c", stale=true Content-Length: 0 <------------> [Mar 10 12:07:52] Scheduling destruction of SIP dialog '55ae664696c0@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:07:52] <--- SIP read from 192.168.1.85:5060 ---> ACK sip:593@24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKq7a080ca15e06daffaee77dc5a87b8385 CSeq: 388021 ACK To: ;tag=as628c2367 Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 From: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec User-Agent: Uniden SIP Phone p2 Ver BS4.77 <-------------> [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: ACK sip:593@24.123.23.170 SIP/2.0 (33) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKq7a080ca15e06daffaee77dc5a87b8385 (82) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 388021 ACK (16) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: ;tag=as628c2367 (42) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 (55) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: From: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec (74) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: (0) [Mar 10 12:07:52] --- (7 headers 0 lines) --- [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received ACK (6) - Command in SIP ACK [Mar 10 12:07:52] <--- SIP read from 192.168.1.83:5060 ---> SIP/2.0 200 OK Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 CSeq: 105 INVITE From: 523 ;tag=as43feeb80 To: 593 ;tag=48204f4e1c15803 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK0140090b Content-Length: 266 Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE Content-Type: application/sdp Supported: replaces Contact: 593 User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 v=0 o=MxSIP 0 1143068287 IN IP4 192.168.1.83 s=SIP Call c=IN IP4 192.168.1.83 t=0 0 m=audio 51000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:on - - - - <-------------> [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: SIP/2.0 200 OK (14) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 (54) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 105 INVITE (16) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: From: 523 ;tag=as43feeb80 (53) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: To: 593 ;tag=48204f4e1c15803 (56) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK0140090b (58) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Content-Length: 266 (19) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE (55) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Content-Type: application/sdp (29) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Supported: replaces (19) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Contact: 593 (40) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 (42) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: (0) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: o=MxSIP 0 1143068287 IN IP4 192.168.1.83 (40) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: s=SIP Call (10) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: c=IN IP4 192.168.1.83 (21) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: m=audio 51000 RTP/AVP 0 8 101 (29) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-15 (15) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=sendrecv (10) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=silenceSupp:on - - - - (24) [Mar 10 12:07:52] --- (12 headers 13 lines) --- [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:2069 __sip_ack: Acked pending invite 105 [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #243 [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on 'a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83' of Request 105: Match Not Found [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:11619 handle_response_invite: SIP response 200 to RE-invite on outgoing call a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 [Mar 10 12:07:52] Found RTP audio format 0 [Mar 10 12:07:52] Found RTP audio format 8 [Mar 10 12:07:52] Found RTP audio format 101 [Mar 10 12:07:52] Peer audio RTP is at port 192.168.1.83:51000 [Mar 10 12:07:52] Found description format PCMU for ID 0 [Mar 10 12:07:52] Found description format PCMA for ID 8 [Mar 10 12:07:52] Found description format telephone-event for ID 101 [Mar 10 12:07:52] Got unsupported a:fmtp in SDP offer [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:5123 process_sdp: T38 state changed to 0 on channel SIP/593-b77052f8 [Mar 10 12:07:52] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) [Mar 10 12:07:52] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 10 12:07:52] Peer audio RTP is at port 192.168.1.83:51000 [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:5203 process_sdp: We're settling with these formats: 0xc (ulaw|alaw) [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:5210 process_sdp: We have an owner, now see if we need to change this call [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:3001 update_call_counter: Updating call counter for incoming call [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:7903 build_route: build_route: Retaining previous route: [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:11752 handle_response_invite: Strange... The other side of the bridge does not have a udptl struct [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:11757 handle_response_invite: T38 state changed to 0 on channel SIP [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:11760 handle_response_invite: T38 state changed to 0 on channel SIP/593-b77052f8 [Mar 10 12:07:52] DEBUG[18555]: chan_sip.c:5637 reqprep: Strict routing enforced for session a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 [Mar 10 12:07:52] set_destination: Parsing for address/port to send to [Mar 10 12:07:52] set_destination: set destination to 192.168.1.83, port 5060 [Mar 10 12:07:52] Transmitting (no NAT) to 192.168.1.83:5060: ACK sip:593@192.168.1.83:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK2a354d38 From: 523 ;tag=as43feeb80 To: 593 ;tag=48204f4e1c15803 Contact: Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 CSeq: 105 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:2008 __sip_autodestruct: Auto destroying SIP dialog '2c01f4fc3cc0317e799f35d0601622e1@24.123.23.170' [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:3107 sip_destroy: Destroying SIP dialog 2c01f4fc3cc0317e799f35d0601622e1@24.123.23.170 [Mar 10 12:07:54] Really destroying SIP dialog '2c01f4fc3cc0317e799f35d0601622e1@24.123.23.170' Method: REGISTER [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:2008 __sip_autodestruct: Auto destroying SIP dialog '000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165' [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:3107 sip_destroy: Destroying SIP dialog 000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165 [Mar 10 12:07:54] Really destroying SIP dialog '000ff78d-ebb20003-688bc9e7-757cda40@192.168.1.165' Method: REGISTER [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:1864 retrans_pkt: SIP TIMER: Rescheduling retransmission #216 (4) SIP/2.0 - 1 [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:1878 retrans_pkt: ** SIP timers: Rescheduling retransmission 5 to 4000 ms (t1 500 ms (Retrans id #216)) [Mar 10 12:07:54] Retransmitting #4 (no NAT) to 192.168.1.85:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKec06c07c3079ca1c8c928ee81120eaffc;received=192.168.1.85 From: 523 ;tag=fb7c7b32eac4985d4d8676d2196e3753 To: ;tag=as59744546 Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 CSeq: 34085 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 18522 18524 IN IP4 192.168.1.97 s=session c=IN IP4 192.168.1.97 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 --- [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:1864 retrans_pkt: SIP TIMER: Rescheduling retransmission #219 (4) SIP/2.0 - 1 [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:1878 retrans_pkt: ** SIP timers: Rescheduling retransmission 5 to 4000 ms (t1 500 ms (Retrans id #219)) [Mar 10 12:07:54] Retransmitting #4 (no NAT) to 192.168.1.85:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKec06c07c3079ca1c8c928ee81120eaffc;received=192.168.1.85 From: 523 ;tag=fb7c7b32eac4985d4d8676d2196e3753 To: ;tag=as59744546 Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 CSeq: 34085 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 18522 18525 IN IP4 192.168.1.97 s=session c=IN IP4 192.168.1.97 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 --- [Mar 10 12:07:54] <--- SIP read from 192.168.1.85:5060 ---> ACK sip:593@24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKq7a080ca15e06daffaee77dc5a87b8385 CSeq: 388021 ACK To: ;tag=as628c2367 Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 From: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec User-Agent: Uniden SIP Phone p2 Ver BS4.77 <-------------> [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: ACK sip:593@24.123.23.170 SIP/2.0 (33) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKq7a080ca15e06daffaee77dc5a87b8385 (82) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 388021 ACK (16) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: ;tag=as628c2367 (42) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 (55) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: From: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec (74) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: (0) [Mar 10 12:07:54] --- (7 headers 0 lines) --- [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received ACK (6) - Command in SIP ACK [Mar 10 12:07:54] <--- 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=z9hG4bKf098d5229453e8cd803d7b2ac98a0f699 Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 48250 REGISTER From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 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="1435817e", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="b9f5b2310a13162083b4809817c40870" <-------------> [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKf098d5229453e8cd803d7b2ac98a0f699 (82) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Call-ID: 55ae66469f7f@24.123.23.170 (35) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: CSeq: 48250 REGISTER (20) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 (66) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="1435817e", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="b9f5b2310a13162083b4809817c40870" (167) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:07:54] --- (13 headers 0 lines) --- [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 48250, ours 48250) [Mar 10 12:07:54] Using latest REGISTER request as basis request [Mar 10 12:07:54] Sending to 192.168.1.93 : 5060 (no NAT) [Mar 10 12:07:54] <--- 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=z9hG4bKf098d5229453e8cd803d7b2ac98a0f699;received=192.168.1.93 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 To: Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 48250 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:54] NOTICE[18555]: chan_sip.c:8135 check_auth: Correct auth, but based on stale nonce received from '' [Mar 10 12:07:54] <--- 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=z9hG4bKf098d5229453e8cd803d7b2ac98a0f699;received=192.168.1.93 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 To: ;tag=as549fb064 Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 48250 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="45b97c87", stale=true Content-Length: 0 <------------> [Mar 10 12:07:54] Scheduling destruction of SIP dialog '55ae66469f7f@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:07:54] <--- SIP read from 192.168.1.83:5060 ---> SIP/2.0 200 OK Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 CSeq: 105 INVITE From: 523 ;tag=as43feeb80 To: 593 ;tag=48204f4e1c15803 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK0140090b Content-Length: 266 Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE Content-Type: application/sdp Supported: replaces Contact: 593 User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 v=0 o=MxSIP 0 1143068287 IN IP4 192.168.1.83 s=SIP Call c=IN IP4 192.168.1.83 t=0 0 m=audio 51000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:on - - - - <-------------> [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: SIP/2.0 200 OK (14) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 (54) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 105 INVITE (16) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: From: 523 ;tag=as43feeb80 (53) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: To: 593 ;tag=48204f4e1c15803 (56) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK0140090b (58) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Content-Length: 266 (19) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE (55) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Content-Type: application/sdp (29) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Supported: replaces (19) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Contact: 593 (40) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 (42) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: (0) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: o=MxSIP 0 1143068287 IN IP4 192.168.1.83 (40) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: s=SIP Call (10) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: c=IN IP4 192.168.1.83 (21) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: m=audio 51000 RTP/AVP 0 8 101 (29) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-15 (15) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=sendrecv (10) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=silenceSupp:on - - - - (24) [Mar 10 12:07:54] --- (12 headers 13 lines) --- [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on 'a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83' of Request 105: Match Found [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:11619 handle_response_invite: SIP response 200 to RE-invite on outgoing call a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 [Mar 10 12:07:54] Found RTP audio format 0 [Mar 10 12:07:54] Found RTP audio format 8 [Mar 10 12:07:54] Found RTP audio format 101 [Mar 10 12:07:54] Peer audio RTP is at port 192.168.1.83:51000 [Mar 10 12:07:54] Found description format PCMU for ID 0 [Mar 10 12:07:54] Found description format PCMA for ID 8 [Mar 10 12:07:54] Found description format telephone-event for ID 101 [Mar 10 12:07:54] Got unsupported a:fmtp in SDP offer [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:5123 process_sdp: T38 state changed to 0 on channel SIP/593-b77052f8 [Mar 10 12:07:54] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) [Mar 10 12:07:54] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 10 12:07:54] Peer audio RTP is at port 192.168.1.83:51000 [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:5203 process_sdp: We're settling with these formats: 0xc (ulaw|alaw) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:5210 process_sdp: We have an owner, now see if we need to change this call [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:3001 update_call_counter: Updating call counter for incoming call [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:7903 build_route: build_route: Retaining previous route: [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:11752 handle_response_invite: Strange... The other side of the bridge does not have a udptl struct [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:11757 handle_response_invite: T38 state changed to 0 on channel SIP [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:11760 handle_response_invite: T38 state changed to 0 on channel SIP/593-b77052f8 [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:5637 reqprep: Strict routing enforced for session a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 [Mar 10 12:07:54] set_destination: Parsing for address/port to send to [Mar 10 12:07:54] set_destination: set destination to 192.168.1.83, port 5060 [Mar 10 12:07:54] Transmitting (no NAT) to 192.168.1.83:5060: ACK sip:593@192.168.1.83:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK4612c2b0 From: 523 ;tag=as43feeb80 To: 593 ;tag=48204f4e1c15803 Contact: Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 CSeq: 105 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 10 12:07:54] <--- 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=z9hG4bKk52a176f208afcf982ef28c96861d4306 Call-ID: 55ae664696c0@24.123.23.170 CSeq: 43041 REGISTER From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 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="1a957140", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="fbea8c836c27273ae975e32edd59f358" <-------------> [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.98:5060;branch=z9hG4bKk52a176f208afcf982ef28c96861d4306 (82) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Call-ID: 55ae664696c0@24.123.23.170 (35) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: CSeq: 43041 REGISTER (20) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 (66) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="1a957140", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="fbea8c836c27273ae975e32edd59f358" (167) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:07:54] --- (13 headers 0 lines) --- [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 43041, ours 43041) [Mar 10 12:07:54] Using latest REGISTER request as basis request [Mar 10 12:07:54] Sending to 192.168.1.98 : 5060 (no NAT) [Mar 10 12:07:54] <--- 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=z9hG4bKk52a176f208afcf982ef28c96861d4306;received=192.168.1.98 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 To: Call-ID: 55ae664696c0@24.123.23.170 CSeq: 43041 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:54] NOTICE[18555]: chan_sip.c:8135 check_auth: Correct auth, but based on stale nonce received from '' [Mar 10 12:07:54] <--- 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=z9hG4bKk52a176f208afcf982ef28c96861d4306;received=192.168.1.98 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 To: ;tag=as08f52ac6 Call-ID: 55ae664696c0@24.123.23.170 CSeq: 43041 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4e53ecf7", stale=true Content-Length: 0 <------------> [Mar 10 12:07:54] Scheduling destruction of SIP dialog '55ae664696c0@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:07:54] <--- SIP read from 192.168.1.83:5060 ---> SIP/2.0 200 OK Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 CSeq: 105 INVITE From: 523 ;tag=as43feeb80 To: 593 ;tag=48204f4e1c15803 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK0140090b Content-Length: 266 Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE Content-Type: application/sdp Supported: replaces Contact: 593 User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 v=0 o=MxSIP 0 1143068287 IN IP4 192.168.1.83 s=SIP Call c=IN IP4 192.168.1.83 t=0 0 m=audio 51000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:on - - - - <-------------> [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: SIP/2.0 200 OK (14) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 (54) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 105 INVITE (16) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: From: 523 ;tag=as43feeb80 (53) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: To: 593 ;tag=48204f4e1c15803 (56) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK0140090b (58) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Content-Length: 266 (19) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE (55) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Content-Type: application/sdp (29) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Supported: replaces (19) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Contact: 593 (40) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 (42) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: (0) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: o=MxSIP 0 1143068287 IN IP4 192.168.1.83 (40) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: s=SIP Call (10) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: c=IN IP4 192.168.1.83 (21) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: m=audio 51000 RTP/AVP 0 8 101 (29) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-15 (15) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=sendrecv (10) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=silenceSupp:on - - - - (24) [Mar 10 12:07:54] --- (12 headers 13 lines) --- [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on 'a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83' of Request 105: Match Found [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:11619 handle_response_invite: SIP response 200 to RE-invite on outgoing call a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 [Mar 10 12:07:54] Found RTP audio format 0 [Mar 10 12:07:54] Found RTP audio format 8 [Mar 10 12:07:54] Found RTP audio format 101 [Mar 10 12:07:54] Peer audio RTP is at port 192.168.1.83:51000 [Mar 10 12:07:54] Found description format PCMU for ID 0 [Mar 10 12:07:54] Found description format PCMA for ID 8 [Mar 10 12:07:54] Found description format telephone-event for ID 101 [Mar 10 12:07:54] Got unsupported a:fmtp in SDP offer [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:5123 process_sdp: T38 state changed to 0 on channel SIP/593-b77052f8 [Mar 10 12:07:54] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) [Mar 10 12:07:54] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 10 12:07:54] Peer audio RTP is at port 192.168.1.83:51000 [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:5203 process_sdp: We're settling with these formats: 0xc (ulaw|alaw) [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:5210 process_sdp: We have an owner, now see if we need to change this call [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:3001 update_call_counter: Updating call counter for incoming call [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:7903 build_route: build_route: Retaining previous route: [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:11752 handle_response_invite: Strange... The other side of the bridge does not have a udptl struct [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:11757 handle_response_invite: T38 state changed to 0 on channel SIP [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:11760 handle_response_invite: T38 state changed to 0 on channel SIP/593-b77052f8 [Mar 10 12:07:54] DEBUG[18555]: chan_sip.c:5637 reqprep: Strict routing enforced for session a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 [Mar 10 12:07:54] set_destination: Parsing for address/port to send to [Mar 10 12:07:54] set_destination: set destination to 192.168.1.83, port 5060 [Mar 10 12:07:54] Transmitting (no NAT) to 192.168.1.83:5060: ACK sip:593@192.168.1.83:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK6e3908f9 From: 523 ;tag=as43feeb80 To: 593 ;tag=48204f4e1c15803 Contact: Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 CSeq: 105 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 10 12:07:54] <--- SIP read from 192.168.1.86:5060 ---> <-------------> [Mar 10 12:07:54] --- (0 headers 0 lines) Nat keepalive --- [Mar 10 12:07:54] <--- 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=z9hG4bKf098d5229453e8cd803d7b2ac98a0f699 Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 48250 REGISTER From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 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="1435817e", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="b9f5b2310a13162083b4809817c40870" <-------------> [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKf098d5229453e8cd803d7b2ac98a0f699 (82) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Call-ID: 55ae66469f7f@24.123.23.170 (35) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: CSeq: 48250 REGISTER (20) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 (66) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="1435817e", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="b9f5b2310a13162083b4809817c40870" (167) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:07:57] --- (13 headers 0 lines) --- [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 48250, ours 48250) [Mar 10 12:07:57] Using latest REGISTER request as basis request [Mar 10 12:07:57] Sending to 192.168.1.93 : 5060 (no NAT) [Mar 10 12:07:57] <--- 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=z9hG4bKf098d5229453e8cd803d7b2ac98a0f699;received=192.168.1.93 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 To: Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 48250 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:57] NOTICE[18555]: chan_sip.c:8135 check_auth: Correct auth, but based on stale nonce received from '' [Mar 10 12:07:57] <--- 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=z9hG4bKf098d5229453e8cd803d7b2ac98a0f699;received=192.168.1.93 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 To: ;tag=as549fb064 Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 48250 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2612775a", stale=true Content-Length: 0 <------------> [Mar 10 12:07:57] Scheduling destruction of SIP dialog '55ae66469f7f@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:1864 retrans_pkt: SIP TIMER: Rescheduling retransmission #239 (4) SIP/2.0 - 1 [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:1878 retrans_pkt: ** SIP timers: Rescheduling retransmission 5 to 4000 ms (t1 500 ms (Retrans id #239)) [Mar 10 12:07:57] Retransmitting #4 (no NAT) to 192.168.1.85:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKz9f560f234760d6220f9cb573a7e43127;received=192.168.1.85 From: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec To: ;tag=as628c2367 Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 388021 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 18522 18526 IN IP4 192.168.1.83 s=session c=IN IP4 192.168.1.83 t=0 0 m=audio 51000 RTP/AVP 0 8 101 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 --- [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:2008 __sip_autodestruct: Auto destroying SIP dialog '000ff78d-ebb20002-23a7f663-5bd85286@192.168.1.165' [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:3107 sip_destroy: Destroying SIP dialog 000ff78d-ebb20002-23a7f663-5bd85286@192.168.1.165 [Mar 10 12:07:57] Really destroying SIP dialog '000ff78d-ebb20002-23a7f663-5bd85286@192.168.1.165' Method: REGISTER [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:1864 retrans_pkt: SIP TIMER: Rescheduling retransmission #216 (5) SIP/2.0 - 1 [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:1878 retrans_pkt: ** SIP timers: Rescheduling retransmission 6 to 4000 ms (t1 500 ms (Retrans id #216)) [Mar 10 12:07:57] Retransmitting #5 (no NAT) to 192.168.1.85:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKec06c07c3079ca1c8c928ee81120eaffc;received=192.168.1.85 From: 523 ;tag=fb7c7b32eac4985d4d8676d2196e3753 To: ;tag=as59744546 Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 CSeq: 34085 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 18522 18524 IN IP4 192.168.1.97 s=session c=IN IP4 192.168.1.97 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 --- [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:1864 retrans_pkt: SIP TIMER: Rescheduling retransmission #219 (5) SIP/2.0 - 1 [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:1878 retrans_pkt: ** SIP timers: Rescheduling retransmission 6 to 4000 ms (t1 500 ms (Retrans id #219)) [Mar 10 12:07:57] Retransmitting #5 (no NAT) to 192.168.1.85:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKec06c07c3079ca1c8c928ee81120eaffc;received=192.168.1.85 From: 523 ;tag=fb7c7b32eac4985d4d8676d2196e3753 To: ;tag=as59744546 Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 CSeq: 34085 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 18522 18525 IN IP4 192.168.1.97 s=session c=IN IP4 192.168.1.97 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 --- [Mar 10 12:07:57] <--- 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=z9hG4bKk52a176f208afcf982ef28c96861d4306 Call-ID: 55ae664696c0@24.123.23.170 CSeq: 43041 REGISTER From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 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="1a957140", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="fbea8c836c27273ae975e32edd59f358" <-------------> [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.98:5060;branch=z9hG4bKk52a176f208afcf982ef28c96861d4306 (82) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Call-ID: 55ae664696c0@24.123.23.170 (35) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: CSeq: 43041 REGISTER (20) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 (66) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="1a957140", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="fbea8c836c27273ae975e32edd59f358" (167) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:07:57] --- (13 headers 0 lines) --- [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 43041, ours 43041) [Mar 10 12:07:57] Using latest REGISTER request as basis request [Mar 10 12:07:57] Sending to 192.168.1.98 : 5060 (no NAT) [Mar 10 12:07:57] <--- 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=z9hG4bKk52a176f208afcf982ef28c96861d4306;received=192.168.1.98 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 To: Call-ID: 55ae664696c0@24.123.23.170 CSeq: 43041 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:57] NOTICE[18555]: chan_sip.c:8135 check_auth: Correct auth, but based on stale nonce received from '' [Mar 10 12:07:57] <--- 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=z9hG4bKk52a176f208afcf982ef28c96861d4306;received=192.168.1.98 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 To: ;tag=as08f52ac6 Call-ID: 55ae664696c0@24.123.23.170 CSeq: 43041 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="022f16af", stale=true Content-Length: 0 <------------> [Mar 10 12:07:57] Scheduling destruction of SIP dialog '55ae664696c0@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:07:57] <--- SIP read from 192.168.1.83:5060 ---> SIP/2.0 200 OK Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 CSeq: 105 INVITE From: 523 ;tag=as43feeb80 To: 593 ;tag=48204f4e1c15803 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK0140090b Content-Length: 266 Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE Content-Type: application/sdp Supported: replaces Contact: 593 User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 v=0 o=MxSIP 0 1143068287 IN IP4 192.168.1.83 s=SIP Call c=IN IP4 192.168.1.83 t=0 0 m=audio 51000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:on - - - - <-------------> [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: SIP/2.0 200 OK (14) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 (54) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 105 INVITE (16) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: From: 523 ;tag=as43feeb80 (53) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: To: 593 ;tag=48204f4e1c15803 (56) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK0140090b (58) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Content-Length: 266 (19) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE (55) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Content-Type: application/sdp (29) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Supported: replaces (19) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Contact: 593 (40) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 (42) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: (0) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: o=MxSIP 0 1143068287 IN IP4 192.168.1.83 (40) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: s=SIP Call (10) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: c=IN IP4 192.168.1.83 (21) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: m=audio 51000 RTP/AVP 0 8 101 (29) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-15 (15) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=sendrecv (10) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=silenceSupp:on - - - - (24) [Mar 10 12:07:57] --- (12 headers 13 lines) --- [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on 'a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83' of Request 105: Match Found [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:11619 handle_response_invite: SIP response 200 to RE-invite on outgoing call a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 [Mar 10 12:07:57] Found RTP audio format 0 [Mar 10 12:07:57] Found RTP audio format 8 [Mar 10 12:07:57] Found RTP audio format 101 [Mar 10 12:07:57] Peer audio RTP is at port 192.168.1.83:51000 [Mar 10 12:07:57] Found description format PCMU for ID 0 [Mar 10 12:07:57] Found description format PCMA for ID 8 [Mar 10 12:07:57] Found description format telephone-event for ID 101 [Mar 10 12:07:57] Got unsupported a:fmtp in SDP offer [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:5123 process_sdp: T38 state changed to 0 on channel SIP/593-b77052f8 [Mar 10 12:07:57] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) [Mar 10 12:07:57] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 10 12:07:57] Peer audio RTP is at port 192.168.1.83:51000 [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:5203 process_sdp: We're settling with these formats: 0xc (ulaw|alaw) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:5210 process_sdp: We have an owner, now see if we need to change this call [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:3001 update_call_counter: Updating call counter for incoming call [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:7903 build_route: build_route: Retaining previous route: [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:11752 handle_response_invite: Strange... The other side of the bridge does not have a udptl struct [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:11757 handle_response_invite: T38 state changed to 0 on channel SIP [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:11760 handle_response_invite: T38 state changed to 0 on channel SIP/593-b77052f8 [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:5637 reqprep: Strict routing enforced for session a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 [Mar 10 12:07:57] set_destination: Parsing for address/port to send to [Mar 10 12:07:57] set_destination: set destination to 192.168.1.83, port 5060 [Mar 10 12:07:57] Transmitting (no NAT) to 192.168.1.83:5060: ACK sip:593@192.168.1.83:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK315478b7 From: 523 ;tag=as43feeb80 To: 593 ;tag=48204f4e1c15803 Contact: Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 CSeq: 105 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 10 12:07:57] <--- SIP read from 192.168.1.83:5060 ---> SIP/2.0 200 OK Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 CSeq: 105 INVITE From: 523 ;tag=as43feeb80 To: 593 ;tag=48204f4e1c15803 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK0140090b Content-Length: 266 Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE Content-Type: application/sdp Supported: replaces Contact: 593 User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 v=0 o=MxSIP 0 1143068287 IN IP4 192.168.1.83 s=SIP Call c=IN IP4 192.168.1.83 t=0 0 m=audio 51000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:on - - - - <-------------> [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: SIP/2.0 200 OK (14) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 (54) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 105 INVITE (16) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: From: 523 ;tag=as43feeb80 (53) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: To: 593 ;tag=48204f4e1c15803 (56) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK0140090b (58) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Content-Length: 266 (19) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE (55) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Content-Type: application/sdp (29) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Supported: replaces (19) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Contact: 593 (40) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 (42) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: (0) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: o=MxSIP 0 1143068287 IN IP4 192.168.1.83 (40) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: s=SIP Call (10) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: c=IN IP4 192.168.1.83 (21) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: m=audio 51000 RTP/AVP 0 8 101 (29) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-15 (15) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=sendrecv (10) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=silenceSupp:on - - - - (24) [Mar 10 12:07:57] --- (12 headers 13 lines) --- [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on 'a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83' of Request 105: Match Found [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:11619 handle_response_invite: SIP response 200 to RE-invite on outgoing call a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 [Mar 10 12:07:57] Found RTP audio format 0 [Mar 10 12:07:57] Found RTP audio format 8 [Mar 10 12:07:57] Found RTP audio format 101 [Mar 10 12:07:57] Peer audio RTP is at port 192.168.1.83:51000 [Mar 10 12:07:57] Found description format PCMU for ID 0 [Mar 10 12:07:57] Found description format PCMA for ID 8 [Mar 10 12:07:57] Found description format telephone-event for ID 101 [Mar 10 12:07:57] Got unsupported a:fmtp in SDP offer [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:5123 process_sdp: T38 state changed to 0 on channel SIP/593-b77052f8 [Mar 10 12:07:57] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) [Mar 10 12:07:57] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 10 12:07:57] Peer audio RTP is at port 192.168.1.83:51000 [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:5203 process_sdp: We're settling with these formats: 0xc (ulaw|alaw) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:5210 process_sdp: We have an owner, now see if we need to change this call [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:3001 update_call_counter: Updating call counter for incoming call [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:7903 build_route: build_route: Retaining previous route: [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:11752 handle_response_invite: Strange... The other side of the bridge does not have a udptl struct [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:11757 handle_response_invite: T38 state changed to 0 on channel SIP [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:11760 handle_response_invite: T38 state changed to 0 on channel SIP/593-b77052f8 [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:5637 reqprep: Strict routing enforced for session a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 [Mar 10 12:07:57] set_destination: Parsing for address/port to send to [Mar 10 12:07:57] set_destination: set destination to 192.168.1.83, port 5060 [Mar 10 12:07:57] Transmitting (no NAT) to 192.168.1.83:5060: ACK sip:593@192.168.1.83:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK53661672 From: 523 ;tag=as43feeb80 To: 593 ;tag=48204f4e1c15803 Contact: Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 CSeq: 105 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 10 12:07:57] <--- 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=z9hG4bKab20528fd38ec686605e8703a298a65fe CSeq: 48251 REGISTER Call-ID: 55ae66469f7f@24.123.23.170 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 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="3ec2039b", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="4eda9906aefe216d14e023a7e85b2cd1" <-------------> [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKab20528fd38ec686605e8703a298a65fe (82) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 48251 REGISTER (20) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Call-ID: 55ae66469f7f@24.123.23.170 (35) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 (66) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="3ec2039b", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="4eda9906aefe216d14e023a7e85b2cd1" (167) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:07:57] --- (13 headers 0 lines) --- [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:07:57] Using latest REGISTER request as basis request [Mar 10 12:07:57] Sending to 192.168.1.93 : 5060 (no NAT) [Mar 10 12:07:57] <--- 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=z9hG4bKab20528fd38ec686605e8703a298a65fe;received=192.168.1.93 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 To: Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 48251 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:57] NOTICE[18555]: chan_sip.c:8135 check_auth: Correct auth, but based on stale nonce received from '' [Mar 10 12:07:57] <--- 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=z9hG4bKab20528fd38ec686605e8703a298a65fe;received=192.168.1.93 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 To: ;tag=as549fb064 Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 48251 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0b686392", stale=true Content-Length: 0 <------------> [Mar 10 12:07:57] Scheduling destruction of SIP dialog '55ae66469f7f@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:07:57] <--- 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=z9hG4bKo7ecad56d8d19a412c257137dfdecda50 CSeq: 43042 REGISTER Call-ID: 55ae664696c0@24.123.23.170 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 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="2e41eec5", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="74e95075c9c9d588ddd40c56af349652" <-------------> [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.98:5060;branch=z9hG4bKo7ecad56d8d19a412c257137dfdecda50 (82) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 43042 REGISTER (20) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Call-ID: 55ae664696c0@24.123.23.170 (35) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 (66) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="2e41eec5", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="74e95075c9c9d588ddd40c56af349652" (167) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:07:57] --- (13 headers 0 lines) --- [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:07:57] Using latest REGISTER request as basis request [Mar 10 12:07:57] Sending to 192.168.1.98 : 5060 (no NAT) [Mar 10 12:07:57] <--- 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=z9hG4bKo7ecad56d8d19a412c257137dfdecda50;received=192.168.1.98 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 To: Call-ID: 55ae664696c0@24.123.23.170 CSeq: 43042 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:57] NOTICE[18555]: chan_sip.c:8135 check_auth: Correct auth, but based on stale nonce received from '' [Mar 10 12:07:57] <--- 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=z9hG4bKo7ecad56d8d19a412c257137dfdecda50;received=192.168.1.98 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 To: ;tag=as08f52ac6 Call-ID: 55ae664696c0@24.123.23.170 CSeq: 43042 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4fa47330", stale=true Content-Length: 0 <------------> [Mar 10 12:07:57] Scheduling destruction of SIP dialog '55ae664696c0@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:07:57] <--- SIP read from 192.168.1.83:5060 ---> BYE sip:523@24.123.23.170 SIP/2.0 Max-Forwards: 70 Content-Length: 0 Via: SIP/2.0/UDP 192.168.1.83:5060;branch=z9hG4bK1f3f69187 Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 From: 593 ;tag=48204f4e1c15803 To: 523 ;tag=as43feeb80 CSeq: 2124963506 BYE Supported: timer Supported: replaces Proxy-Authorization:Digest response="a25f229b4140f22fd8f81735a64a9273",username="593",realm="asterisk",nonce="5f611baa",algorithm=MD5,uri="sip:523@24.123.23.170" User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 <-------------> [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: BYE sip:523@24.123.23.170 SIP/2.0 (33) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Max-Forwards: 70 (16) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Content-Length: 0 (17) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Via: SIP/2.0/UDP 192.168.1.83:5060;branch=z9hG4bK1f3f69187 (58) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 (54) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: From: 593 ;tag=48204f4e1c15803 (58) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: To: 523 ;tag=as43feeb80 (51) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: CSeq: 2124963506 BYE (20) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Supported: timer (16) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Supported: replaces (19) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Proxy-Authorization:Digest response="a25f229b4140f22fd8f81735a64a9273",username="593",realm="asterisk",nonce="5f611baa",algorithm=MD5,uri="sip:523@24.123.23.170" (161) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 (42) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: (0) [Mar 10 12:07:57] --- (12 headers 0 lines) --- [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received BYE (8) - Command in SIP BYE [Mar 10 12:07:57] Sending to 192.168.1.83 : 5060 (no NAT) [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:1631 sip_alreadygone: Setting SIP_ALREADYGONE on dialog a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 [Mar 10 12:07:57] DEBUG[18555]: chan_sip.c:14167 handle_request_bye: Received bye, issuing owner hangup [Mar 10 12:07:57] <--- Transmitting (no NAT) to 192.168.1.83:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.83:5060;branch=z9hG4bK1f3f69187;received=192.168.1.83 From: 593 ;tag=48204f4e1c15803 To: 523 ;tag=as43feeb80 Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 CSeq: 2124963506 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:07:57] DEBUG[18591]: rtp.c:2855 bridge_native_loop: Oooh, got a hangup [Mar 10 12:07:57] DEBUG[18591]: chan_sip.c:16860 sip_set_rtp_peer: Sending reinvite on SIP '53c067b5491f609931bc7a0965b73fd3@24.123.23.170' - It's audio soon redirected to IP 24.123.23.170 [Mar 10 12:07:57] DEBUG[18591]: chan_sip.c:5637 reqprep: Strict routing enforced for session 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 [Mar 10 12:07:57] set_destination: Parsing for address/port to send to [Mar 10 12:07:57] set_destination: set destination to 192.168.1.85, port 5060 [Mar 10 12:07:57] DEBUG[18591]: chan_sip.c:6182 add_sdp: ** Our capability: 0xc (ulaw|alaw) Video flag: True [Mar 10 12:07:57] DEBUG[18591]: chan_sip.c:6183 add_sdp: ** Our prefcodec: 0x4 (ulaw) [Mar 10 12:07:57] Audio is at 24.123.23.170 port 10848 [Mar 10 12:07:57] Adding codec 0x4 (ulaw) to SDP [Mar 10 12:07:57] Adding codec 0x8 (alaw) to SDP [Mar 10 12:07:57] Adding non-codec 0x1 (telephone-event) to SDP [Mar 10 12:07:57] DEBUG[18591]: chan_sip.c:6314 add_sdp: -- Done with adding codecs to SDP [Mar 10 12:07:57] DEBUG[18591]: chan_sip.c:6359 add_sdp: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Mar 10 12:07:57] DEBUG[18591]: chan_sip.c:1619 initialize_initreq: Initializing already initialized SIP dialog 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 (presumably reinvite) [Mar 10 12:07:57] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 0: INVITE sip:523@192.168.1.85:5060 SIP/2.0 (40) [Mar 10 12:07:57] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK01745cf2 (58) [Mar 10 12:07:57] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 2: From: ;tag=as628c2367 (44) [Mar 10 12:07:57] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 3: To: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec (72) [Mar 10 12:07:59] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 4: Contact: (32) [Mar 10 12:07:59] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 5: Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 (55) [Mar 10 12:07:59] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 6: CSeq: 105 INVITE (16) [Mar 10 12:07:59] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Mar 10 12:07:59] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:07:59] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Mar 10 12:07:59] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 10: Supported: replaces (19) [Mar 10 12:07:59] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Mar 10 12:07:59] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 12: Content-Type: application/sdp (29) [Mar 10 12:07:59] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 13: Content-Length: 266 (19) [Mar 10 12:07:59] DEBUG[18591]: chan_sip.c:4571 parse_request: Header 14: (0) [Mar 10 12:07:59] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:07:59] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: o=root 18522 18528 IN IP4 24.123.23.170 (39) [Mar 10 12:07:59] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: s=session (9) [Mar 10 12:07:59] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: c=IN IP4 24.123.23.170 (22) [Mar 10 12:07:59] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:07:59] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: m=audio 10848 RTP/AVP 0 8 101 (29) [Mar 10 12:08:00] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:08:00] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Mar 10 12:08:00] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:08:00] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-16 (15) [Mar 10 12:08:00] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=silenceSupp:off - - - - (25) [Mar 10 12:08:00] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:08:00] DEBUG[18591]: chan_sip.c:4603 parse_request: Line: a=sendrecv (10) [Mar 10 12:08:00] Reliably Transmitting (no NAT) to 192.168.1.85:5060: INVITE sip:523@192.168.1.85:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK01745cf2 From: ;tag=as628c2367 To: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec Contact: Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 105 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 18522 18528 IN IP4 24.123.23.170 s=session c=IN IP4 24.123.23.170 t=0 0 m=audio 10848 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 --- [Mar 10 12:08:00] DEBUG[18591]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #254 [Mar 10 12:08:00] DEBUG[18591]: channel.c:4048 ast_channel_bridge: Returning from native bridge, channels: SIP/593-b77052f8, SIP/523-087dd9a8 [Mar 10 12:08:00] <--- 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=z9hG4bKab20528fd38ec686605e8703a298a65fe CSeq: 48251 REGISTER Call-ID: 55ae66469f7f@24.123.23.170 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 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="3ec2039b", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="4eda9906aefe216d14e023a7e85b2cd1" <-------------> [Mar 10 12:08:00] DEBUG[18591]: channel.c:1693 ast_hangup: Hanging up channel 'SIP/523-087dd9a8' [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKab20528fd38ec686605e8703a298a65fe (82) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 48251 REGISTER (20) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Call-ID: 55ae66469f7f@24.123.23.170 (35) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 (66) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="3ec2039b", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="4eda9906aefe216d14e023a7e85b2cd1" (167) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:08:00] --- (13 headers 0 lines) --- [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 48251, ours 48251) [Mar 10 12:08:00] Using latest REGISTER request as basis request [Mar 10 12:08:00] Sending to 192.168.1.93 : 5060 (no NAT) [Mar 10 12:08:00] <--- 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=z9hG4bKab20528fd38ec686605e8703a298a65fe;received=192.168.1.93 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 To: Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 48251 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:08:00] NOTICE[18555]: chan_sip.c:8135 check_auth: Correct auth, but based on stale nonce received from '' [Mar 10 12:08:00] <--- 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=z9hG4bKab20528fd38ec686605e8703a298a65fe;received=192.168.1.93 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 To: ;tag=as549fb064 Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 48251 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5010491a", stale=true Content-Length: 0 <------------> [Mar 10 12:08:00] Scheduling destruction of SIP dialog '55ae66469f7f@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:2008 __sip_autodestruct: Auto destroying SIP dialog '1319586744@192.168.1.97' [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:3107 sip_destroy: Destroying SIP dialog 1319586744@192.168.1.97 [Mar 10 12:08:00] Really destroying SIP dialog '1319586744@192.168.1.97' Method: REGISTER [Mar 10 12:08:00] DEBUG[18591]: chan_sip.c:3310 sip_hangup: Hangup call SIP/523-087dd9a8, SIP callid 53c067b5491f609931bc7a0965b73fd3@24.123.23.170) [Mar 10 12:08:00] Scheduling destruction of SIP dialog '53c067b5491f609931bc7a0965b73fd3@24.123.23.170' in 32000 ms (Method: ACK) [Mar 10 12:08:00] DEBUG[18591]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/523-087dd9a8 [Mar 10 12:08:00] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 523 [Mar 10 12:08:00] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 523 [Mar 10 12:08:00] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/523 - state 1 (Not in use) [Mar 10 12:08:00] DEBUG[18591]: rtp.c:1474 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Mar 10 12:08:00] DEBUG[18591]: app_dial.c:1670 dial_exec_full: Exiting with DIALSTATUS=ANSWER. [Mar 10 12:08:00] DEBUG[18591]: pbx.c:2389 __ast_pbx_run: Spawn extension (smvoice-sip,523,6) exited non-zero on 'SIP/593-b77052f8' [Mar 10 12:08:00] == Spawn extension (smvoice-sip, 523, 6) exited non-zero on 'SIP/593-b77052f8' [Mar 10 12:08:00] DEBUG[18591]: pbx.c:1644 pbx_substitute_variables_helper_full: Function result is '"593 593" <593>' [Mar 10 12:08:00] DEBUG[18591]: pbx.c:1644 pbx_substitute_variables_helper_full: Function result is '593' [Mar 10 12:08:00] DEBUG[18591]: pbx.c:1644 pbx_substitute_variables_helper_full: Function result is '523' [Mar 10 12:08:00] DEBUG[18591]: pbx.c:1644 pbx_substitute_variables_helper_full: Function result is 'smvoice-sip' [Mar 10 12:08:00] DEBUG[18591]: pbx.c:1644 pbx_substitute_variables_helper_full: Function result is 'SIP/593-b77052f8' [Mar 10 12:08:00] DEBUG[18591]: pbx.c:1644 pbx_substitute_variables_helper_full: Function result is 'SIP/523-087dd9a8' [Mar 10 12:08:00] DEBUG[18591]: pbx.c:1644 pbx_substitute_variables_helper_full: Function result is 'Dial' [Mar 10 12:08:00] DEBUG[18591]: pbx.c:1644 pbx_substitute_variables_helper_full: Function result is 'SIP/523|20|' [Mar 10 12:08:00] DEBUG[18591]: pbx.c:1644 pbx_substitute_variables_helper_full: Function result is '2007-03-10 12:07:26' [Mar 10 12:08:00] DEBUG[18591]: pbx.c:1644 pbx_substitute_variables_helper_full: Function result is '2007-03-10 12:07:29' [Mar 10 12:08:00] DEBUG[18591]: pbx.c:1644 pbx_substitute_variables_helper_full: Function result is '2007-03-10 12:08:00' [Mar 10 12:08:00] DEBUG[18591]: pbx.c:1644 pbx_substitute_variables_helper_full: Function result is '34' [Mar 10 12:08:00] DEBUG[18591]: pbx.c:1644 pbx_substitute_variables_helper_full: Function result is '31' [Mar 10 12:08:00] DEBUG[18591]: pbx.c:1644 pbx_substitute_variables_helper_full: Function result is 'ANSWERED' [Mar 10 12:08:00] DEBUG[18591]: pbx.c:1644 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' [Mar 10 12:08:00] DEBUG[18591]: pbx.c:1644 pbx_substitute_variables_helper_full: Function result is '' [Mar 10 12:08:00] DEBUG[18591]: pbx.c:1644 pbx_substitute_variables_helper_full: Function result is '1173546446.0' [Mar 10 12:08:00] DEBUG[18591]: pbx.c:1644 pbx_substitute_variables_helper_full: Function result is '' [Mar 10 12:08:00] DEBUG[18591]: channel.c:1693 ast_hangup: Hanging up channel 'SIP/593-b77052f8' [Mar 10 12:08:00] DEBUG[18591]: chan_sip.c:3310 sip_hangup: Hangup call SIP/593-b77052f8, SIP callid a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83) [Mar 10 12:08:00] DEBUG[18591]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/593-b77052f8 [Mar 10 12:08:00] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 593 [Mar 10 12:08:00] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 593 [Mar 10 12:08:00] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/593 - state 1 (Not in use) [Mar 10 12:08:00] DEBUG[18622]: 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. [Mar 10 12:08:00] DEBUG[18623]: app_queue.c:546 changethread: Device 'SIP/593' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Mar 10 12:08:00] DEBUG[18555]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/528 [Mar 10 12:08:00] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 528 [Mar 10 12:08:00] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 528 [Mar 10 12:08:00] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/528 - state 5 (Unavailable) [Mar 10 12:08:00] DEBUG[18555]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/529 [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:1864 retrans_pkt: SIP TIMER: Rescheduling retransmission #239 (5) SIP/2.0 - 1 [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:1878 retrans_pkt: ** SIP timers: Rescheduling retransmission 6 to 4000 ms (t1 500 ms (Retrans id #239)) [Mar 10 12:08:00] Retransmitting #5 (no NAT) to 192.168.1.85:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKz9f560f234760d6220f9cb573a7e43127;received=192.168.1.85 From: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec To: ;tag=as628c2367 Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 388021 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 18522 18526 IN IP4 192.168.1.83 s=session c=IN IP4 192.168.1.83 t=0 0 m=audio 51000 RTP/AVP 0 8 101 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 --- [Mar 10 12:08:00] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 529 [Mar 10 12:08:00] DEBUG[18527]: chan_sip.c:15201 sip_devicestate: Checking device state for peer 529 [Mar 10 12:08:00] Really destroying SIP dialog 'a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83' Method: BYE [Mar 10 12:08:00] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for SIP/529 - state 5 (Unavailable) [Mar 10 12:08:00] <--- 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=z9hG4bKo7ecad56d8d19a412c257137dfdecda50 CSeq: 43042 REGISTER Call-ID: 55ae664696c0@24.123.23.170 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 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="2e41eec5", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="74e95075c9c9d588ddd40c56af349652" <-------------> [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.98:5060;branch=z9hG4bKo7ecad56d8d19a412c257137dfdecda50 (82) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 43042 REGISTER (20) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Call-ID: 55ae664696c0@24.123.23.170 (35) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 (66) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="2e41eec5", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="74e95075c9c9d588ddd40c56af349652" (167) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:08:00] --- (13 headers 0 lines) --- [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 43042, ours 43042) [Mar 10 12:08:00] Using latest REGISTER request as basis request [Mar 10 12:08:00] Sending to 192.168.1.98 : 5060 (no NAT) [Mar 10 12:08:00] <--- 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=z9hG4bKo7ecad56d8d19a412c257137dfdecda50;received=192.168.1.98 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 To: Call-ID: 55ae664696c0@24.123.23.170 CSeq: 43042 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:08:00] NOTICE[18555]: chan_sip.c:8135 check_auth: Correct auth, but based on stale nonce received from '' [Mar 10 12:08:00] <--- 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=z9hG4bKo7ecad56d8d19a412c257137dfdecda50;received=192.168.1.98 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 To: ;tag=as08f52ac6 Call-ID: 55ae664696c0@24.123.23.170 CSeq: 43042 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1e7808cf", stale=true Content-Length: 0 <------------> [Mar 10 12:08:00] Scheduling destruction of SIP dialog '55ae664696c0@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:08:00] <--- SIP read from 192.168.1.83:5060 ---> BYE sip:523@24.123.23.170 SIP/2.0 Max-Forwards: 70 Content-Length: 0 Via: SIP/2.0/UDP 192.168.1.83:5060;branch=z9hG4bK1f3f69187 Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 From: 593 ;tag=48204f4e1c15803 To: 523 ;tag=as43feeb80 CSeq: 2124963506 BYE Supported: timer Supported: replaces Proxy-Authorization:Digest response="a25f229b4140f22fd8f81735a64a9273",username="593",realm="asterisk",nonce="5f611baa",algorithm=MD5,uri="sip:523@24.123.23.170" User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 <-------------> [Mar 10 12:08:00] DEBUG[18625]: app_queue.c:546 changethread: Device 'SIP/529' changed to state '5' (Unavailable) but we don't care because they're not a member of any queue. [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: BYE sip:523@24.123.23.170 SIP/2.0 (33) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Max-Forwards: 70 (16) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Content-Length: 0 (17) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Via: SIP/2.0/UDP 192.168.1.83:5060;branch=z9hG4bK1f3f69187 (58) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 (54) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: From: 593 ;tag=48204f4e1c15803 (58) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: To: 523 ;tag=as43feeb80 (51) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: CSeq: 2124963506 BYE (20) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Supported: timer (16) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Supported: replaces (19) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Proxy-Authorization:Digest response="a25f229b4140f22fd8f81735a64a9273",username="593",realm="asterisk",nonce="5f611baa",algorithm=MD5,uri="sip:523@24.123.23.170" (161) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 (42) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: (0) [Mar 10 12:08:00] --- (12 headers 0 lines) --- [Mar 10 12:08:00] <--- Transmitting (no NAT) to 192.168.1.83:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 192.168.1.83:5060;branch=z9hG4bK1f3f69187;received=192.168.1.83 From: 593 ;tag=48204f4e1c15803 To: 523 ;tag=as43feeb80 Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 CSeq: 2124963506 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:14774 sipsock_read: Invalid SIP message - rejected , no callid, len 571 [Mar 10 12:08:00] <--- 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=z9hG4bKab20528fd38ec686605e8703a298a65fe CSeq: 48251 REGISTER Call-ID: 55ae66469f7f@24.123.23.170 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 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="3ec2039b", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="4eda9906aefe216d14e023a7e85b2cd1" <-------------> [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKab20528fd38ec686605e8703a298a65fe (82) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 48251 REGISTER (20) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Call-ID: 55ae66469f7f@24.123.23.170 (35) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 (66) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="3ec2039b", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="4eda9906aefe216d14e023a7e85b2cd1" (167) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:08:00] --- (13 headers 0 lines) --- [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 48251, ours 48251) [Mar 10 12:08:00] Using latest REGISTER request as basis request [Mar 10 12:08:00] Sending to 192.168.1.93 : 5060 (no NAT) [Mar 10 12:08:00] DEBUG[18624]: app_queue.c:546 changethread: Device 'SIP/528' changed to state '5' (Unavailable) but we don't care because they're not a member of any queue. [Mar 10 12:08:00] <--- 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=z9hG4bKab20528fd38ec686605e8703a298a65fe;received=192.168.1.93 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 To: Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 48251 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:08:00] NOTICE[18555]: chan_sip.c:8135 check_auth: Correct auth, but based on stale nonce received from '' [Mar 10 12:08:00] <--- 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=z9hG4bKab20528fd38ec686605e8703a298a65fe;received=192.168.1.93 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 To: ;tag=as549fb064 Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 48251 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1ba05bdd", stale=true Content-Length: 0 <------------> [Mar 10 12:08:00] Scheduling destruction of SIP dialog '55ae66469f7f@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:08:00] <--- 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=z9hG4bKo7ecad56d8d19a412c257137dfdecda50 CSeq: 43042 REGISTER Call-ID: 55ae664696c0@24.123.23.170 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 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="2e41eec5", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="74e95075c9c9d588ddd40c56af349652" <-------------> [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.98:5060;branch=z9hG4bKo7ecad56d8d19a412c257137dfdecda50 (82) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 43042 REGISTER (20) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Call-ID: 55ae664696c0@24.123.23.170 (35) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 (66) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="2e41eec5", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="74e95075c9c9d588ddd40c56af349652" (167) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:08:00] --- (13 headers 0 lines) --- [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 43042, ours 43042) [Mar 10 12:08:00] Using latest REGISTER request as basis request [Mar 10 12:08:00] Sending to 192.168.1.98 : 5060 (no NAT) [Mar 10 12:08:00] <--- 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=z9hG4bKo7ecad56d8d19a412c257137dfdecda50;received=192.168.1.98 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 To: Call-ID: 55ae664696c0@24.123.23.170 CSeq: 43042 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:08:00] NOTICE[18555]: chan_sip.c:8135 check_auth: Correct auth, but based on stale nonce received from '' [Mar 10 12:08:00] <--- 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=z9hG4bKo7ecad56d8d19a412c257137dfdecda50;received=192.168.1.98 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 To: ;tag=as08f52ac6 Call-ID: 55ae664696c0@24.123.23.170 CSeq: 43042 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="45915593", stale=true Content-Length: 0 <------------> [Mar 10 12:08:00] Scheduling destruction of SIP dialog '55ae664696c0@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:08:00] <--- SIP read from 192.168.1.83:5060 ---> BYE sip:523@24.123.23.170 SIP/2.0 Max-Forwards: 70 Content-Length: 0 Via: SIP/2.0/UDP 192.168.1.83:5060;branch=z9hG4bK1f3f69187 Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 From: 593 ;tag=48204f4e1c15803 To: 523 ;tag=as43feeb80 CSeq: 2124963506 BYE Supported: timer Supported: replaces Proxy-Authorization:Digest response="a25f229b4140f22fd8f81735a64a9273",username="593",realm="asterisk",nonce="5f611baa",algorithm=MD5,uri="sip:523@24.123.23.170" User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 <-------------> [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: BYE sip:523@24.123.23.170 SIP/2.0 (33) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Max-Forwards: 70 (16) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Content-Length: 0 (17) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Via: SIP/2.0/UDP 192.168.1.83:5060;branch=z9hG4bK1f3f69187 (58) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 (54) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: From: 593 ;tag=48204f4e1c15803 (58) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: To: 523 ;tag=as43feeb80 (51) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: CSeq: 2124963506 BYE (20) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Supported: timer (16) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Supported: replaces (19) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Proxy-Authorization:Digest response="a25f229b4140f22fd8f81735a64a9273",username="593",realm="asterisk",nonce="5f611baa",algorithm=MD5,uri="sip:523@24.123.23.170" (161) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: User-Agent: Callctrl/1.6.0.0 MxSF/v3.5.3.4 (42) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: (0) [Mar 10 12:08:00] --- (12 headers 0 lines) --- [Mar 10 12:08:00] <--- Transmitting (no NAT) to 192.168.1.83:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 192.168.1.83:5060;branch=z9hG4bK1f3f69187;received=192.168.1.83 From: 593 ;tag=48204f4e1c15803 To: 523 ;tag=as43feeb80 Call-ID: a6aff3f0c46b60e8a002e109fa1da338@192.168.1.83 CSeq: 2124963506 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:14774 sipsock_read: Invalid SIP message - rejected , no callid, len 571 [Mar 10 12:08:00] <--- 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=z9hG4bKab20528fd38ec686605e8703a298a65fe CSeq: 48251 REGISTER Call-ID: 55ae66469f7f@24.123.23.170 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 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="3ec2039b", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="4eda9906aefe216d14e023a7e85b2cd1" <-------------> [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKab20528fd38ec686605e8703a298a65fe (82) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 48251 REGISTER (20) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Call-ID: 55ae66469f7f@24.123.23.170 (35) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 (66) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="3ec2039b", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="4eda9906aefe216d14e023a7e85b2cd1" (167) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:08:00] --- (13 headers 0 lines) --- [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 48251, ours 48251) [Mar 10 12:08:00] Using latest REGISTER request as basis request [Mar 10 12:08:00] Sending to 192.168.1.93 : 5060 (no NAT) [Mar 10 12:08:00] <--- 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=z9hG4bKab20528fd38ec686605e8703a298a65fe;received=192.168.1.93 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 To: Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 48251 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:08:00] NOTICE[18555]: chan_sip.c:8135 check_auth: Correct auth, but based on stale nonce received from '' [Mar 10 12:08:00] <--- 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=z9hG4bKab20528fd38ec686605e8703a298a65fe;received=192.168.1.93 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 To: ;tag=as549fb064 Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 48251 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="29574d03", stale=true Content-Length: 0 <------------> [Mar 10 12:08:00] Scheduling destruction of SIP dialog '55ae66469f7f@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:08:00] <--- 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=z9hG4bKo7ecad56d8d19a412c257137dfdecda50 CSeq: 43042 REGISTER Call-ID: 55ae664696c0@24.123.23.170 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 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="2e41eec5", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="74e95075c9c9d588ddd40c56af349652" <-------------> [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.98:5060;branch=z9hG4bKo7ecad56d8d19a412c257137dfdecda50 (82) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 43042 REGISTER (20) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Call-ID: 55ae664696c0@24.123.23.170 (35) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 (66) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="2e41eec5", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="74e95075c9c9d588ddd40c56af349652" (167) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:08:00] --- (13 headers 0 lines) --- [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 43042, ours 43042) [Mar 10 12:08:00] Using latest REGISTER request as basis request [Mar 10 12:08:00] Sending to 192.168.1.98 : 5060 (no NAT) [Mar 10 12:08:00] <--- 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=z9hG4bKo7ecad56d8d19a412c257137dfdecda50;received=192.168.1.98 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 To: Call-ID: 55ae664696c0@24.123.23.170 CSeq: 43042 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:08:00] NOTICE[18555]: chan_sip.c:8135 check_auth: Correct auth, but based on stale nonce received from '' [Mar 10 12:08:00] <--- 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=z9hG4bKo7ecad56d8d19a412c257137dfdecda50;received=192.168.1.98 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 To: ;tag=as08f52ac6 Call-ID: 55ae664696c0@24.123.23.170 CSeq: 43042 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7db35a24", stale=true Content-Length: 0 <------------> [Mar 10 12:08:00] Scheduling destruction of SIP dialog '55ae664696c0@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:08:00] <--- 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=z9hG4bKq083eb526770846fd5a4541e353374aad CSeq: 48252 REGISTER Call-ID: 55ae66469f7f@24.123.23.170 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 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="0b686392", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="e65a3b61f594b27ebbb14445870d2b14" <-------------> [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKq083eb526770846fd5a4541e353374aad (82) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 48252 REGISTER (20) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Call-ID: 55ae66469f7f@24.123.23.170 (35) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 (66) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="0b686392", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="e65a3b61f594b27ebbb14445870d2b14" (167) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:08:00] --- (13 headers 0 lines) --- [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:08:00] Using latest REGISTER request as basis request [Mar 10 12:08:00] Sending to 192.168.1.93 : 5060 (no NAT) [Mar 10 12:08:00] <--- 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=z9hG4bKq083eb526770846fd5a4541e353374aad;received=192.168.1.93 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 To: Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 48252 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:08:00] NOTICE[18555]: chan_sip.c:8135 check_auth: Correct auth, but based on stale nonce received from '' [Mar 10 12:08:00] <--- 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=z9hG4bKq083eb526770846fd5a4541e353374aad;received=192.168.1.93 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 To: ;tag=as549fb064 Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 48252 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="37147f29", stale=true Content-Length: 0 <------------> [Mar 10 12:08:00] Scheduling destruction of SIP dialog '55ae66469f7f@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:08:00] <--- 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=z9hG4bKa9243d20c7ffc624577d9f752d5af0dc2 CSeq: 43043 REGISTER Call-ID: 55ae664696c0@24.123.23.170 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 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="4fa47330", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="94a24636c6d633d8eb3df2fb4a3cd7ae" <-------------> [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.98:5060;branch=z9hG4bKa9243d20c7ffc624577d9f752d5af0dc2 (82) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 43043 REGISTER (20) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Call-ID: 55ae664696c0@24.123.23.170 (35) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 (66) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:00] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="4fa47330", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="94a24636c6d633d8eb3df2fb4a3cd7ae" (167) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:08:02] --- (13 headers 0 lines) --- [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:08:02] Using latest REGISTER request as basis request [Mar 10 12:08:02] Sending to 192.168.1.98 : 5060 (no NAT) [Mar 10 12:08:02] <--- 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=z9hG4bKa9243d20c7ffc624577d9f752d5af0dc2;received=192.168.1.98 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 To: Call-ID: 55ae664696c0@24.123.23.170 CSeq: 43043 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:08:02] NOTICE[18555]: chan_sip.c:8135 check_auth: Correct auth, but based on stale nonce received from '' [Mar 10 12:08:02] <--- 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=z9hG4bKa9243d20c7ffc624577d9f752d5af0dc2;received=192.168.1.98 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 To: ;tag=as08f52ac6 Call-ID: 55ae664696c0@24.123.23.170 CSeq: 43043 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="00273f8d", stale=true Content-Length: 0 <------------> [Mar 10 12:08:02] Scheduling destruction of SIP dialog '55ae664696c0@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:1864 retrans_pkt: SIP TIMER: Rescheduling retransmission #254 (1) INVITE - 5 [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:1878 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #254)) [Mar 10 12:08:02] Retransmitting #1 (no NAT) to 192.168.1.85:5060: INVITE sip:523@192.168.1.85:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK01745cf2 From: ;tag=as628c2367 To: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec Contact: Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 105 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 18522 18528 IN IP4 24.123.23.170 s=session c=IN IP4 24.123.23.170 t=0 0 m=audio 10848 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 --- [Mar 10 12:08:02] DEBUG[18555]: sched.c:204 sched_settime: Request to schedule in the past?!?! [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:1864 retrans_pkt: SIP TIMER: Rescheduling retransmission #216 (6) SIP/2.0 - 1 [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:1878 retrans_pkt: ** SIP timers: Rescheduling retransmission 7 to 4000 ms (t1 500 ms (Retrans id #216)) [Mar 10 12:08:02] Retransmitting #6 (no NAT) to 192.168.1.85:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKec06c07c3079ca1c8c928ee81120eaffc;received=192.168.1.85 From: 523 ;tag=fb7c7b32eac4985d4d8676d2196e3753 To: ;tag=as59744546 Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 CSeq: 34085 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 18522 18524 IN IP4 192.168.1.97 s=session c=IN IP4 192.168.1.97 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 --- [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:1864 retrans_pkt: SIP TIMER: Rescheduling retransmission #219 (6) SIP/2.0 - 1 [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:1878 retrans_pkt: ** SIP timers: Rescheduling retransmission 7 to 4000 ms (t1 500 ms (Retrans id #219)) [Mar 10 12:08:02] Retransmitting #6 (no NAT) to 192.168.1.85:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKec06c07c3079ca1c8c928ee81120eaffc;received=192.168.1.85 From: 523 ;tag=fb7c7b32eac4985d4d8676d2196e3753 To: ;tag=as59744546 Call-ID: 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 CSeq: 34085 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 18522 18525 IN IP4 192.168.1.97 s=session c=IN IP4 192.168.1.97 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 --- [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:1864 retrans_pkt: SIP TIMER: Rescheduling retransmission #254 (2) INVITE - 5 [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:1878 retrans_pkt: ** SIP timers: Rescheduling retransmission 3 to 2000 ms (t1 500 ms (Retrans id #254)) [Mar 10 12:08:02] Retransmitting #2 (no NAT) to 192.168.1.85:5060: INVITE sip:523@192.168.1.85:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK01745cf2 From: ;tag=as628c2367 To: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec Contact: Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 105 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 18522 18528 IN IP4 24.123.23.170 s=session c=IN IP4 24.123.23.170 t=0 0 m=audio 10848 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 --- [Mar 10 12:08:02] <--- SIP read from 192.168.1.97:5060 ---> <-------------> [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: (0) [Mar 10 12:08:02] --- (0 headers 0 lines) Nat keepalive --- [Mar 10 12:08:02] <--- 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=z9hG4bKa9243d20c7ffc624577d9f752d5af0dc2 CSeq: 43043 REGISTER Call-ID: 55ae664696c0@24.123.23.170 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 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="4fa47330", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="94a24636c6d633d8eb3df2fb4a3cd7ae" <-------------> [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.98:5060;branch=z9hG4bKa9243d20c7ffc624577d9f752d5af0dc2 (82) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 43043 REGISTER (20) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Call-ID: 55ae664696c0@24.123.23.170 (35) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 (66) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="4fa47330", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="94a24636c6d633d8eb3df2fb4a3cd7ae" (167) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:08:02] --- (13 headers 0 lines) --- [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 43043, ours 43043) [Mar 10 12:08:02] Using latest REGISTER request as basis request [Mar 10 12:08:02] Sending to 192.168.1.98 : 5060 (no NAT) [Mar 10 12:08:02] <--- 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=z9hG4bKa9243d20c7ffc624577d9f752d5af0dc2;received=192.168.1.98 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 To: Call-ID: 55ae664696c0@24.123.23.170 CSeq: 43043 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:08:02] NOTICE[18555]: chan_sip.c:8135 check_auth: Correct auth, but based on stale nonce received from '' [Mar 10 12:08:02] <--- 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=z9hG4bKa9243d20c7ffc624577d9f752d5af0dc2;received=192.168.1.98 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 To: ;tag=as08f52ac6 Call-ID: 55ae664696c0@24.123.23.170 CSeq: 43043 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7f42c0ef", stale=true Content-Length: 0 <------------> [Mar 10 12:08:02] Scheduling destruction of SIP dialog '55ae664696c0@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:08:02] <--- 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=z9hG4bKq083eb526770846fd5a4541e353374aad CSeq: 48252 REGISTER Call-ID: 55ae66469f7f@24.123.23.170 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 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="0b686392", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="e65a3b61f594b27ebbb14445870d2b14" <-------------> [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKq083eb526770846fd5a4541e353374aad (82) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 48252 REGISTER (20) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Call-ID: 55ae66469f7f@24.123.23.170 (35) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 (66) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="0b686392", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="e65a3b61f594b27ebbb14445870d2b14" (167) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:08:02] --- (13 headers 0 lines) --- [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 48252, ours 48252) [Mar 10 12:08:02] Using latest REGISTER request as basis request [Mar 10 12:08:02] Sending to 192.168.1.93 : 5060 (no NAT) [Mar 10 12:08:02] <--- 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=z9hG4bKq083eb526770846fd5a4541e353374aad;received=192.168.1.93 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 To: Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 48252 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:08:02] NOTICE[18555]: chan_sip.c:8135 check_auth: Correct auth, but based on stale nonce received from '' [Mar 10 12:08:02] <--- 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=z9hG4bKq083eb526770846fd5a4541e353374aad;received=192.168.1.93 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 To: ;tag=as549fb064 Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 48252 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="53d9d24f", stale=true Content-Length: 0 <------------> [Mar 10 12:08:02] Scheduling destruction of SIP dialog '55ae66469f7f@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:08:02] <--- 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=z9hG4bKz5f3ecc157cf1ea8b88f4e3639df84824 Call-ID: 55ae66476793@24.123.23.170 CSeq: 39011 REGISTER From: ;tag=6726509f7fef65bf50359d83fd756f9f 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="41de4491", algorithm=MD5, uri="sip:24.123.23.170", username="525", response="4f06ff263d3a173a88252a09d4bf3350" <-------------> [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bKz5f3ecc157cf1ea8b88f4e3639df84824 (82) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Call-ID: 55ae66476793@24.123.23.170 (35) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: CSeq: 39011 REGISTER (20) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=6726509f7fef65bf50359d83fd756f9f (66) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="41de4491", algorithm=MD5, uri="sip:24.123.23.170", username="525", response="4f06ff263d3a173a88252a09d4bf3350" (167) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:08:02] --- (13 headers 0 lines) --- [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4308 sip_alloc: Allocating new SIP dialog for 55ae66476793@24.123.23.170 - REGISTER (No RTP) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:08:02] Using latest REGISTER request as basis request [Mar 10 12:08:02] Sending to 192.168.1.76 : 5060 (no NAT) [Mar 10 12:08:02] <--- 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=z9hG4bKz5f3ecc157cf1ea8b88f4e3639df84824;received=192.168.1.76 From: ;tag=6726509f7fef65bf50359d83fd756f9f To: Call-ID: 55ae66476793@24.123.23.170 CSeq: 39011 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:08:02] <--- 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=z9hG4bKz5f3ecc157cf1ea8b88f4e3639df84824;received=192.168.1.76 From: ;tag=6726509f7fef65bf50359d83fd756f9f To: ;tag=as182bf0b1 Call-ID: 55ae66476793@24.123.23.170 CSeq: 39011 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="23304027" Content-Length: 0 <------------> [Mar 10 12:08:02] Scheduling destruction of SIP dialog '55ae66476793@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:08:02] <--- 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=z9hG4bKa9243d20c7ffc624577d9f752d5af0dc2 CSeq: 43043 REGISTER Call-ID: 55ae664696c0@24.123.23.170 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 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="4fa47330", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="94a24636c6d633d8eb3df2fb4a3cd7ae" <-------------> [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.98:5060;branch=z9hG4bKa9243d20c7ffc624577d9f752d5af0dc2 (82) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 43043 REGISTER (20) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Call-ID: 55ae664696c0@24.123.23.170 (35) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 (66) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="4fa47330", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="94a24636c6d633d8eb3df2fb4a3cd7ae" (167) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:08:02] --- (13 headers 0 lines) --- [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 43043, ours 43043) [Mar 10 12:08:02] Using latest REGISTER request as basis request [Mar 10 12:08:02] Sending to 192.168.1.98 : 5060 (no NAT) [Mar 10 12:08:02] <--- 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=z9hG4bKa9243d20c7ffc624577d9f752d5af0dc2;received=192.168.1.98 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 To: Call-ID: 55ae664696c0@24.123.23.170 CSeq: 43043 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:08:02] NOTICE[18555]: chan_sip.c:8135 check_auth: Correct auth, but based on stale nonce received from '' [Mar 10 12:08:02] <--- 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=z9hG4bKa9243d20c7ffc624577d9f752d5af0dc2;received=192.168.1.98 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 To: ;tag=as08f52ac6 Call-ID: 55ae664696c0@24.123.23.170 CSeq: 43043 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="35568fd7", stale=true Content-Length: 0 <------------> [Mar 10 12:08:02] Scheduling destruction of SIP dialog '55ae664696c0@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:08:02] <--- 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=z9hG4bKq083eb526770846fd5a4541e353374aad CSeq: 48252 REGISTER Call-ID: 55ae66469f7f@24.123.23.170 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 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="0b686392", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="e65a3b61f594b27ebbb14445870d2b14" <-------------> [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKq083eb526770846fd5a4541e353374aad (82) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 48252 REGISTER (20) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Call-ID: 55ae66469f7f@24.123.23.170 (35) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 (66) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="0b686392", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="e65a3b61f594b27ebbb14445870d2b14" (167) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:08:02] --- (13 headers 0 lines) --- [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 48252, ours 48252) [Mar 10 12:08:02] Using latest REGISTER request as basis request [Mar 10 12:08:02] Sending to 192.168.1.93 : 5060 (no NAT) [Mar 10 12:08:02] <--- 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=z9hG4bKq083eb526770846fd5a4541e353374aad;received=192.168.1.93 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 To: Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 48252 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:08:02] NOTICE[18555]: chan_sip.c:8135 check_auth: Correct auth, but based on stale nonce received from '' [Mar 10 12:08:02] <--- 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=z9hG4bKq083eb526770846fd5a4541e353374aad;received=192.168.1.93 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 To: ;tag=as549fb064 Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 48252 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="03f42a02", stale=true Content-Length: 0 <------------> [Mar 10 12:08:02] Scheduling destruction of SIP dialog '55ae66469f7f@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:08:02] <--- 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=z9hG4bKz5f3ecc157cf1ea8b88f4e3639df84824 Call-ID: 55ae66476793@24.123.23.170 CSeq: 39011 REGISTER From: ;tag=6726509f7fef65bf50359d83fd756f9f 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="41de4491", algorithm=MD5, uri="sip:24.123.23.170", username="525", response="4f06ff263d3a173a88252a09d4bf3350" <-------------> [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bKz5f3ecc157cf1ea8b88f4e3639df84824 (82) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Call-ID: 55ae66476793@24.123.23.170 (35) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: CSeq: 39011 REGISTER (20) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=6726509f7fef65bf50359d83fd756f9f (66) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="41de4491", algorithm=MD5, uri="sip:24.123.23.170", username="525", response="4f06ff263d3a173a88252a09d4bf3350" (167) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:08:02] --- (13 headers 0 lines) --- [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 39011, ours 39011) [Mar 10 12:08:02] Using latest REGISTER request as basis request [Mar 10 12:08:02] Sending to 192.168.1.76 : 5060 (no NAT) [Mar 10 12:08:02] <--- 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=z9hG4bKz5f3ecc157cf1ea8b88f4e3639df84824;received=192.168.1.76 From: ;tag=6726509f7fef65bf50359d83fd756f9f To: Call-ID: 55ae66476793@24.123.23.170 CSeq: 39011 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:08:02] NOTICE[18555]: chan_sip.c:8135 check_auth: Correct auth, but based on stale nonce received from '' [Mar 10 12:08:02] <--- 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=z9hG4bKz5f3ecc157cf1ea8b88f4e3639df84824;received=192.168.1.76 From: ;tag=6726509f7fef65bf50359d83fd756f9f To: ;tag=as182bf0b1 Call-ID: 55ae66476793@24.123.23.170 CSeq: 39011 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5893da66", stale=true Content-Length: 0 <------------> [Mar 10 12:08:02] Scheduling destruction of SIP dialog '55ae66476793@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:08:02] <--- SIP read from 192.168.1.85:5060 ---> SIP/2.0 200 OK To: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec Contact: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK01745cf2 From: ;tag=as628c2367 Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 105 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=- 18522 204409 IN IP4 192.168.1.85 s=session c=IN IP4 192.168.1.85 t=0 0 m=audio 30016 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 <-------------> [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: SIP/2.0 200 OK (14) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: To: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec (72) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Contact: (36) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK01745cf2 (58) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=as628c2367 (44) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 (55) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: CSeq: 105 INVITE (16) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Content-Type: application/sdp (29) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Content-Length: 247 (19) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: (0) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: o=- 18522 204409 IN IP4 192.168.1.85 (36) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: s=session (9) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: c=IN IP4 192.168.1.85 (21) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: m=audio 30016 RTP/AVP 0 101 (27) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-15,16,17,18 (24) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=silenceSupp:off - - - - (25) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=sendrecv (10) [Mar 10 12:08:02] --- (11 headers 12 lines) --- [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:2069 __sip_ack: Acked pending invite 105 [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #254 [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '53c067b5491f609931bc7a0965b73fd3@24.123.23.170' of Request 105: Match Not Found [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:11621 handle_response_invite: SIP response 200 to standard invite [Mar 10 12:08:02] Found RTP audio format 0 [Mar 10 12:08:02] Found RTP audio format 101 [Mar 10 12:08:02] Peer audio RTP is at port 192.168.1.85:30016 [Mar 10 12:08:02] Found description format PCMU for ID 0 [Mar 10 12:08:02] Found description format telephone-event for ID 101 [Mar 10 12:08:02] Got unsupported a:fmtp in SDP offer [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:5123 process_sdp: T38 state changed to 0 on channel [Mar 10 12:08:02] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 10 12:08:02] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 10 12:08:02] Peer audio RTP is at port 192.168.1.85:30016 [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:5203 process_sdp: We're settling with these formats: 0x4 (ulaw) [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:3001 update_call_counter: Updating call counter for outgoing call [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:7903 build_route: build_route: Retaining previous route: [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:5637 reqprep: Strict routing enforced for session 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 [Mar 10 12:08:02] set_destination: Parsing for address/port to send to [Mar 10 12:08:02] set_destination: set destination to 192.168.1.85, port 5060 [Mar 10 12:08:02] Transmitting (no NAT) to 192.168.1.85:5060: ACK sip:523@192.168.1.85:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK0dc9c835 From: ;tag=as628c2367 To: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec Contact: Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 105 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:5637 reqprep: Strict routing enforced for session 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 [Mar 10 12:08:02] set_destination: Parsing for address/port to send to [Mar 10 12:08:02] set_destination: set destination to 192.168.1.85, port 5060 [Mar 10 12:08:02] Reliably Transmitting (no NAT) to 192.168.1.85:5060: BYE sip:523@192.168.1.85:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5bdd0182 From: ;tag=as628c2367 To: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 106 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 10 12:08:02] DEBUG[18555]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #270 [Mar 10 12:08:02] Scheduling destruction of SIP dialog '53c067b5491f609931bc7a0965b73fd3@24.123.23.170' in 32000 ms (Method: ACK) [Mar 10 12:08:02] <--- 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=z9hG4bKk9ce3370c8525e6b0e8c0a1cb0bd7ae04 CSeq: 48253 REGISTER Call-ID: 55ae66469f7f@24.123.23.170 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 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="37147f29", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="9ea7effb31c722bbcec8b04fb74a36e1" <-------------> [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKk9ce3370c8525e6b0e8c0a1cb0bd7ae04 (82) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 48253 REGISTER (20) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Call-ID: 55ae66469f7f@24.123.23.170 (35) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 (66) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="37147f29", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="9ea7effb31c722bbcec8b04fb74a36e1" (167) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:08:04] --- (13 headers 0 lines) --- [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:08:04] Using latest REGISTER request as basis request [Mar 10 12:08:04] Sending to 192.168.1.93 : 5060 (no NAT) [Mar 10 12:08:04] <--- 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=z9hG4bKk9ce3370c8525e6b0e8c0a1cb0bd7ae04;received=192.168.1.93 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 To: Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 48253 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:08:04] NOTICE[18555]: chan_sip.c:8135 check_auth: Correct auth, but based on stale nonce received from '' [Mar 10 12:08:04] <--- 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=z9hG4bKk9ce3370c8525e6b0e8c0a1cb0bd7ae04;received=192.168.1.93 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 To: ;tag=as549fb064 Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 48253 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4579536e", stale=true Content-Length: 0 <------------> [Mar 10 12:08:04] Scheduling destruction of SIP dialog '55ae66469f7f@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:1864 retrans_pkt: SIP TIMER: Rescheduling retransmission #270 (1) BYE - 8 [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:1878 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #270)) [Mar 10 12:08:04] Retransmitting #1 (no NAT) to 192.168.1.85:5060: BYE sip:523@192.168.1.85:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5bdd0182 From: ;tag=as628c2367 To: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 106 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 10 12:08:04] DEBUG[18555]: sched.c:204 sched_settime: Request to schedule in the past?!?! [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:1864 retrans_pkt: SIP TIMER: Rescheduling retransmission #239 (6) SIP/2.0 - 1 [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:1878 retrans_pkt: ** SIP timers: Rescheduling retransmission 7 to 4000 ms (t1 500 ms (Retrans id #239)) [Mar 10 12:08:04] Retransmitting #6 (no NAT) to 192.168.1.85:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKz9f560f234760d6220f9cb573a7e43127;received=192.168.1.85 From: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec To: ;tag=as628c2367 Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 388021 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 18522 18526 IN IP4 192.168.1.83 s=session c=IN IP4 192.168.1.83 t=0 0 m=audio 51000 RTP/AVP 0 8 101 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 --- [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:1864 retrans_pkt: SIP TIMER: Rescheduling retransmission #270 (2) BYE - 8 [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:1878 retrans_pkt: ** SIP timers: Rescheduling retransmission 3 to 2000 ms (t1 500 ms (Retrans id #270)) [Mar 10 12:08:04] Retransmitting #2 (no NAT) to 192.168.1.85:5060: BYE sip:523@192.168.1.85:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5bdd0182 From: ;tag=as628c2367 To: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 106 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 10 12:08:04] <--- 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=z9hG4bKz5f3ecc157cf1ea8b88f4e3639df84824 Call-ID: 55ae66476793@24.123.23.170 CSeq: 39011 REGISTER From: ;tag=6726509f7fef65bf50359d83fd756f9f 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="41de4491", algorithm=MD5, uri="sip:24.123.23.170", username="525", response="4f06ff263d3a173a88252a09d4bf3350" <-------------> [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bKz5f3ecc157cf1ea8b88f4e3639df84824 (82) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Call-ID: 55ae66476793@24.123.23.170 (35) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: CSeq: 39011 REGISTER (20) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=6726509f7fef65bf50359d83fd756f9f (66) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="41de4491", algorithm=MD5, uri="sip:24.123.23.170", username="525", response="4f06ff263d3a173a88252a09d4bf3350" (167) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:08:04] --- (13 headers 0 lines) --- [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 39011, ours 39011) [Mar 10 12:08:04] Using latest REGISTER request as basis request [Mar 10 12:08:04] Sending to 192.168.1.76 : 5060 (no NAT) [Mar 10 12:08:04] <--- 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=z9hG4bKz5f3ecc157cf1ea8b88f4e3639df84824;received=192.168.1.76 From: ;tag=6726509f7fef65bf50359d83fd756f9f To: Call-ID: 55ae66476793@24.123.23.170 CSeq: 39011 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:08:04] NOTICE[18555]: chan_sip.c:8135 check_auth: Correct auth, but based on stale nonce received from '' [Mar 10 12:08:04] <--- 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=z9hG4bKz5f3ecc157cf1ea8b88f4e3639df84824;received=192.168.1.76 From: ;tag=6726509f7fef65bf50359d83fd756f9f To: ;tag=as182bf0b1 Call-ID: 55ae66476793@24.123.23.170 CSeq: 39011 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="58df7a9c", stale=true Content-Length: 0 <------------> [Mar 10 12:08:04] Scheduling destruction of SIP dialog '55ae66476793@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:08:04] <--- SIP read from 192.168.1.85:5060 ---> SIP/2.0 200 OK To: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec Contact: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK01745cf2 From: ;tag=as628c2367 Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 105 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=- 18522 204409 IN IP4 192.168.1.85 s=session c=IN IP4 192.168.1.85 t=0 0 m=audio 30016 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 <-------------> [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: SIP/2.0 200 OK (14) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: To: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec (72) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Contact: (36) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK01745cf2 (58) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=as628c2367 (44) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 (55) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: CSeq: 105 INVITE (16) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Content-Type: application/sdp (29) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Content-Length: 247 (19) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: (0) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: o=- 18522 204409 IN IP4 192.168.1.85 (36) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: s=session (9) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: c=IN IP4 192.168.1.85 (21) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: m=audio 30016 RTP/AVP 0 101 (27) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-15,16,17,18 (24) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=silenceSupp:off - - - - (25) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=sendrecv (10) [Mar 10 12:08:04] --- (11 headers 12 lines) --- [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '53c067b5491f609931bc7a0965b73fd3@24.123.23.170' of Request 105: Match Found [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:11621 handle_response_invite: SIP response 200 to standard invite [Mar 10 12:08:04] Found RTP audio format 0 [Mar 10 12:08:04] Found RTP audio format 101 [Mar 10 12:08:04] Peer audio RTP is at port 192.168.1.85:30016 [Mar 10 12:08:04] Found description format PCMU for ID 0 [Mar 10 12:08:04] Found description format telephone-event for ID 101 [Mar 10 12:08:04] Got unsupported a:fmtp in SDP offer [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:5123 process_sdp: T38 state changed to 0 on channel [Mar 10 12:08:04] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 10 12:08:04] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 10 12:08:04] Peer audio RTP is at port 192.168.1.85:30016 [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:5203 process_sdp: We're settling with these formats: 0x4 (ulaw) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:3001 update_call_counter: Updating call counter for outgoing call [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:7903 build_route: build_route: Retaining previous route: [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:5637 reqprep: Strict routing enforced for session 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 [Mar 10 12:08:04] set_destination: Parsing for address/port to send to [Mar 10 12:08:04] set_destination: set destination to 192.168.1.85, port 5060 [Mar 10 12:08:04] Transmitting (no NAT) to 192.168.1.85:5060: ACK sip:523@192.168.1.85:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK31ab3a01 From: ;tag=as628c2367 To: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec Contact: Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 105 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 10 12:08:04] <--- 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=z9hG4bKk9ce3370c8525e6b0e8c0a1cb0bd7ae04 CSeq: 48253 REGISTER Call-ID: 55ae66469f7f@24.123.23.170 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 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="37147f29", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="9ea7effb31c722bbcec8b04fb74a36e1" <-------------> [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKk9ce3370c8525e6b0e8c0a1cb0bd7ae04 (82) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 48253 REGISTER (20) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Call-ID: 55ae66469f7f@24.123.23.170 (35) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 (66) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="37147f29", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="9ea7effb31c722bbcec8b04fb74a36e1" (167) [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:08:04] --- (13 headers 0 lines) --- [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:08:04] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 48253, ours 48253) [Mar 10 12:08:04] Using latest REGISTER request as basis request [Mar 10 12:08:04] Sending to 192.168.1.93 : 5060 (no NAT) [Mar 10 12:08:04] <--- 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=z9hG4bKk9ce3370c8525e6b0e8c0a1cb0bd7ae04;received=192.168.1.93 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 To: Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 48253 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:08:04] NOTICE[18555]: chan_sip.c:8135 check_auth: Correct auth, but based on stale nonce received from '' [Mar 10 12:08:04] <--- 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=z9hG4bKk9ce3370c8525e6b0e8c0a1cb0bd7ae04;received=192.168.1.93 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 To: ;tag=as549fb064 Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 48253 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3c6891ea", stale=true Content-Length: 0 <------------> [Mar 10 12:08:04] Scheduling destruction of SIP dialog '55ae66469f7f@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:08:04] <--- SIP read from 68.58.36.157:5060 ---> <-------------> [Mar 10 12:08:04] --- (0 headers 0 lines) Nat keepalive --- [Mar 10 12:08:04] <--- 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=z9hG4bKa9243d20c7ffc624577d9f752d5af0dc2 CSeq: 43043 REGISTER Call-ID: 55ae664696c0@24.123.23.170 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 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="4fa47330", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="94a24636c6d633d8eb3df2fb4a3cd7ae" <-------------> [Mar 10 12:08:06] DEBUG[18548]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel IAX2/606 [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.98:5060;branch=z9hG4bKa9243d20c7ffc624577d9f752d5af0dc2 (82) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 43043 REGISTER (20) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Call-ID: 55ae664696c0@24.123.23.170 (35) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 (66) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="4fa47330", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="94a24636c6d633d8eb3df2fb4a3cd7ae" (167) stop[Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:08:07] --- (13 headers 0 lines) --- [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 43043, ours 43043) [Mar 10 12:08:07] Using latest REGISTER request as basis request [Mar 10 12:08:07] Sending to 192.168.1.98 : 5060 (no NAT) [Mar 10 12:08:07] <--- 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=z9hG4bKa9243d20c7ffc624577d9f752d5af0dc2;received=192.168.1.98 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 To: Call-ID: 55ae664696c0@24.123.23.170 CSeq: 43043 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:08:07] NOTICE[18555]: chan_sip.c:8135 check_auth: Correct auth, but based on stale nonce received from '' [Mar 10 12:08:07] <--- 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=z9hG4bKa9243d20c7ffc624577d9f752d5af0dc2;received=192.168.1.98 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 To: ;tag=as08f52ac6 Call-ID: 55ae664696c0@24.123.23.170 CSeq: 43043 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="44b14db6", stale=true Content-Length: 0 <------------> [Mar 10 12:08:07] Scheduling destruction of SIP dialog '55ae664696c0@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:08:07] WARNING[18555]: chan_sip.c:1897 retrans_pkt: Maximum retries exceeded on transmission 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 for seqno 34085 (Critical Response) [Mar 10 12:08:07] WARNING[18555]: chan_sip.c:1897 retrans_pkt: Maximum retries exceeded on transmission 6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170 for seqno 34085 (Critical Response) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:2008 __sip_autodestruct: Auto destroying SIP dialog '55ae6647678c@24.123.23.170' [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:3107 sip_destroy: Destroying SIP dialog 55ae6647678c@24.123.23.170 [Mar 10 12:08:07] Really destroying SIP dialog '55ae6647678c@24.123.23.170' Method: REGISTER [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:1864 retrans_pkt: SIP TIMER: Rescheduling retransmission #270 (3) BYE - 8 [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:1878 retrans_pkt: ** SIP timers: Rescheduling retransmission 4 to 4000 ms (t1 500 ms (Retrans id #270)) [Mar 10 12:08:07] Retransmitting #3 (no NAT) to 192.168.1.85:5060: BYE sip:523@192.168.1.85:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5bdd0182 From: ;tag=as628c2367 To: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 106 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 10 12:08:07] WARNING[18555]: chan_sip.c:1897 retrans_pkt: Maximum retries exceeded on transmission 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 for seqno 388021 (Critical Response) [Mar 10 12:08:07] Really destroying SIP dialog '6e16fb8411c8b8a460d03f1f39371fa2@24.123.23.170' Method: ACK [Mar 10 12:08:07] <--- SIP read from 192.168.1.86:5062 ---> <-------------> [Mar 10 12:08:07] --- (0 headers 0 lines) Nat keepalive --- [Mar 10 12:08:07] <--- SIP read from 192.168.1.85:5060 ---> BYE sip:593@24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKi2034d1cd4bddbaf9eab46925ade753a8 Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 388022 BYE From: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec To: ;tag=as628c2367 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 <-------------> [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: BYE sip:593@24.123.23.170 SIP/2.0 (33) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKi2034d1cd4bddbaf9eab46925ade753a8 (82) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 (55) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: CSeq: 388022 BYE (16) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec (74) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: ;tag=as628c2367 (42) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Max-Forwards: 70 (16) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Content-Length: 0 (17) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: (0) [Mar 10 12:08:07] --- (11 headers 0 lines) --- [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received BYE (8) - Command in SIP BYE [Mar 10 12:08:07] Sending to 192.168.1.85 : 5060 (no NAT) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:1631 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 [Mar 10 12:08:07] Scheduling destruction of SIP dialog '53c067b5491f609931bc7a0965b73fd3@24.123.23.170' in 32000 ms (Method: BYE) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:14171 handle_request_bye: Received bye, no owner, selfdestruct soon. [Mar 10 12:08:07] <--- 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=z9hG4bKi2034d1cd4bddbaf9eab46925ade753a8;received=192.168.1.85 From: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec To: ;tag=as628c2367 Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 388022 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:08:07] <--- SIP read from 192.168.1.85:5060 ---> SIP/2.0 200 OK To: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec Contact: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK01745cf2 From: ;tag=as628c2367 Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 105 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=- 18522 204409 IN IP4 192.168.1.85 s=session c=IN IP4 192.168.1.85 t=0 0 m=audio 30016 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 <-------------> [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: SIP/2.0 200 OK (14) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: To: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec (72) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Contact: (36) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK01745cf2 (58) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=as628c2367 (44) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 (55) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: CSeq: 105 INVITE (16) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Content-Type: application/sdp (29) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Content-Length: 247 (19) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: (0) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: o=- 18522 204409 IN IP4 192.168.1.85 (36) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: s=session (9) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: c=IN IP4 192.168.1.85 (21) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: m=audio 30016 RTP/AVP 0 101 (27) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-15,16,17,18 (24) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=silenceSupp:off - - - - (25) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=sendrecv (10) [Mar 10 12:08:07] --- (11 headers 12 lines) --- [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '53c067b5491f609931bc7a0965b73fd3@24.123.23.170' of Request 105: Match Found [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:11621 handle_response_invite: SIP response 200 to standard invite [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:11626 handle_response_invite: Got response on call that is already terminated: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 (ignoring) [Mar 10 12:08:07] <--- 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=z9hG4bKk9ce3370c8525e6b0e8c0a1cb0bd7ae04 CSeq: 48253 REGISTER Call-ID: 55ae66469f7f@24.123.23.170 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 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="37147f29", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="9ea7effb31c722bbcec8b04fb74a36e1" <-------------> [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKk9ce3370c8525e6b0e8c0a1cb0bd7ae04 (82) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 48253 REGISTER (20) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Call-ID: 55ae66469f7f@24.123.23.170 (35) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 (66) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="37147f29", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="9ea7effb31c722bbcec8b04fb74a36e1" (167) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:08:07] --- (13 headers 0 lines) --- [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 48253, ours 48253) [Mar 10 12:08:07] Using latest REGISTER request as basis request [Mar 10 12:08:07] Sending to 192.168.1.93 : 5060 (no NAT) [Mar 10 12:08:07] <--- 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=z9hG4bKk9ce3370c8525e6b0e8c0a1cb0bd7ae04;received=192.168.1.93 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 To: Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 48253 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:08:07] NOTICE[18555]: chan_sip.c:8135 check_auth: Correct auth, but based on stale nonce received from '' [Mar 10 12:08:07] <--- 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=z9hG4bKk9ce3370c8525e6b0e8c0a1cb0bd7ae04;received=192.168.1.93 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 To: ;tag=as549fb064 Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 48253 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="63c2d8ed", stale=true Content-Length: 0 <------------> [Mar 10 12:08:07] Scheduling destruction of SIP dialog '55ae66469f7f@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:08:07] <--- SIP read from 192.168.1.85:5060 ---> BYE sip:593@24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKi2034d1cd4bddbaf9eab46925ade753a8 Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 388022 BYE From: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec To: ;tag=as628c2367 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 <-------------> [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: BYE sip:593@24.123.23.170 SIP/2.0 (33) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKi2034d1cd4bddbaf9eab46925ade753a8 (82) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 (55) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: CSeq: 388022 BYE (16) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec (74) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: ;tag=as628c2367 (42) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Max-Forwards: 70 (16) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Content-Length: 0 (17) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: (0) [Mar 10 12:08:07] --- (11 headers 0 lines) --- [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received BYE (8) - Command in SIP BYE [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (BYE Seqno 388022, ours 388022) [Mar 10 12:08:07] Sending to 192.168.1.85 : 5060 (no NAT) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:1631 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 [Mar 10 12:08:07] Scheduling destruction of SIP dialog '53c067b5491f609931bc7a0965b73fd3@24.123.23.170' in 32000 ms (Method: BYE) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:14171 handle_request_bye: Received bye, no owner, selfdestruct soon. [Mar 10 12:08:07] <--- 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=z9hG4bKi2034d1cd4bddbaf9eab46925ade753a8;received=192.168.1.85 From: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec To: ;tag=as628c2367 Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 388022 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:08:07] <--- SIP read from 192.168.1.85:5060 ---> SIP/2.0 200 OK To: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec Contact: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK01745cf2 From: ;tag=as628c2367 Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 105 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=- 18522 204409 IN IP4 192.168.1.85 s=session c=IN IP4 192.168.1.85 t=0 0 m=audio 30016 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 <-------------> [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: SIP/2.0 200 OK (14) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: To: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec (72) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Contact: (36) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK01745cf2 (58) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=as628c2367 (44) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 (55) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: CSeq: 105 INVITE (16) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Content-Type: application/sdp (29) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Content-Length: 247 (19) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:07] DEBUG[18547]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel IAX2/606 [Mar 10 12:08:07] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for IAX2 - 606 [Mar 10 12:08:07] DEBUG[18527]: chan_iax2.c:9636 iax2_devicestate: Checking device state for device 606 [Mar 10 12:08:07] DEBUG[18527]: chan_iax2.c:9644 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 606? addr=-1509840704, defaddr=0 maxms=0, lastms=0 [Mar 10 12:08:07] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for IAX2/606 - state 1 (Not in use) [Mar 10 12:08:07] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for IAX2 - 606 [Mar 10 12:08:07] DEBUG[18527]: chan_iax2.c:9636 iax2_devicestate: Checking device state for device 606 [Mar 10 12:08:07] DEBUG[18527]: chan_iax2.c:9644 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 606? addr=-1509840704, defaddr=0 maxms=0, lastms=0 [Mar 10 12:08:07] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for IAX2/606 - state 1 (Not in use) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: (0) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: o=- 18522 204409 IN IP4 192.168.1.85 (36) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: s=session (9) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: c=IN IP4 192.168.1.85 (21) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: m=audio 30016 RTP/AVP 0 101 (27) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-15,16,17,18 (24) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=silenceSupp:off - - - - (25) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=sendrecv (10) [Mar 10 12:08:07] --- (11 headers 12 lines) --- [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '53c067b5491f609931bc7a0965b73fd3@24.123.23.170' of Request 105: Match Found [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:11621 handle_response_invite: SIP response 200 to standard invite [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:11626 handle_response_invite: Got response on call that is already terminated: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 (ignoring) [Mar 10 12:08:07] <--- 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=z9hG4bKpe1c089c552f0fcea2227e46bd5cd0765 CSeq: 43044 REGISTER Call-ID: 55ae664696c0@24.123.23.170 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 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="00273f8d", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="adf9acf187d165830272bcbd77f67cb3" <-------------> [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.98:5060;branch=z9hG4bKpe1c089c552f0fcea2227e46bd5cd0765 (82) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 43044 REGISTER (20) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Call-ID: 55ae664696c0@24.123.23.170 (35) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 (66) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="00273f8d", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="adf9acf187d165830272bcbd77f67cb3" (167) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:08:07] --- (13 headers 0 lines) --- [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:08:07] Using latest REGISTER request as basis request [Mar 10 12:08:07] Sending to 192.168.1.98 : 5060 (no NAT) [Mar 10 12:08:07] <--- 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=z9hG4bKpe1c089c552f0fcea2227e46bd5cd0765;received=192.168.1.98 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 To: Call-ID: 55ae664696c0@24.123.23.170 CSeq: 43044 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:08:07] NOTICE[18555]: chan_sip.c:8135 check_auth: Correct auth, but based on stale nonce received from '' [Mar 10 12:08:07] <--- 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=z9hG4bKpe1c089c552f0fcea2227e46bd5cd0765;received=192.168.1.98 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 To: ;tag=as08f52ac6 Call-ID: 55ae664696c0@24.123.23.170 CSeq: 43044 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="297c5574", stale=true Content-Length: 0 <------------> [Mar 10 12:08:07] Scheduling destruction of SIP dialog '55ae664696c0@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:08:07] <--- 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=z9hG4bKve765c9620c30adb0386fbe1fadcc8616 CSeq: 39012 REGISTER Call-ID: 55ae66476793@24.123.23.170 From: ;tag=6726509f7fef65bf50359d83fd756f9f 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="23304027", algorithm=MD5, uri="sip:24.123.23.170", username="525", response="b0d75dbb247e3827e8c2fb498ab2edc3" <-------------> [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bKve765c9620c30adb0386fbe1fadcc8616 (82) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 39012 REGISTER (20) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Call-ID: 55ae66476793@24.123.23.170 (35) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=6726509f7fef65bf50359d83fd756f9f (66) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:08:07] DEBUG[18627]: 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. [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:07] DEBUG[18628]: 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. [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="23304027", algorithm=MD5, uri="sip:24.123.23.170", username="525", response="b0d75dbb247e3827e8c2fb498ab2edc3" (167) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:08:07] --- (13 headers 0 lines) --- [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:08:07] Using latest REGISTER request as basis request [Mar 10 12:08:07] Sending to 192.168.1.76 : 5060 (no NAT) [Mar 10 12:08:07] <--- 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=z9hG4bKve765c9620c30adb0386fbe1fadcc8616;received=192.168.1.76 From: ;tag=6726509f7fef65bf50359d83fd756f9f To: Call-ID: 55ae66476793@24.123.23.170 CSeq: 39012 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:08:07] NOTICE[18555]: chan_sip.c:8135 check_auth: Correct auth, but based on stale nonce received from '' [Mar 10 12:08:07] DEBUG[18545]: chan_iax2.c:7327 socket_process: Immediately destroying 4, having received INVAL [Mar 10 12:08:07] <--- 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=z9hG4bKve765c9620c30adb0386fbe1fadcc8616;received=192.168.1.76 From: ;tag=6726509f7fef65bf50359d83fd756f9f To: ;tag=as182bf0b1 Call-ID: 55ae66476793@24.123.23.170 CSeq: 39012 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0db85705", stale=true Content-Length: 0 <------------> [Mar 10 12:08:07] DEBUG[18545]: chan_iax2.c:7330 socket_process: Destroying call 4 [Mar 10 12:08:07] Scheduling destruction of SIP dialog '55ae66476793@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:08:07] <--- SIP read from 192.168.1.85:5060 ---> SIP/2.0 200 OK To: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec Contact: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK01745cf2 From: ;tag=as628c2367 Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 105 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=- 18522 204409 IN IP4 192.168.1.85 s=session c=IN IP4 192.168.1.85 t=0 0 m=audio 30016 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 <-------------> [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: SIP/2.0 200 OK (14) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: To: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec (72) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Contact: (36) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK01745cf2 (58) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=as628c2367 (44) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 (55) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: CSeq: 105 INVITE (16) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Content-Type: application/sdp (29) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Content-Length: 247 (19) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: (0) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: v=0 (3) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: o=- 18522 204409 IN IP4 192.168.1.85 (36) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: s=session (9) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: c=IN IP4 192.168.1.85 (21) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: t=0 0 (5) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: m=audio 30016 RTP/AVP 0 101 (27) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=fmtp:101 0-15,16,17,18 (24) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=silenceSupp:off - - - - (25) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=ptime:20 (10) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4603 parse_request: Line: a=sendrecv (10) [Mar 10 12:08:07] --- (11 headers 12 lines) --- [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '53c067b5491f609931bc7a0965b73fd3@24.123.23.170' of Request 105: Match Found [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:11621 handle_response_invite: SIP response 200 to standard invite [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:11626 handle_response_invite: Got response on call that is already terminated: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 (ignoring) [Mar 10 12:08:07] <--- SIP read from 192.168.1.85:5060 ---> SIP/2.0 200 OK To: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5bdd0182 From: ;tag=as628c2367 Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 106 BYE Content-Length: 0 Supported: sip-cc, sip-cc-01, replaces, timer Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK <-------------> [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: SIP/2.0 200 OK (14) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: To: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec (72) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5bdd0182 (58) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: From: ;tag=as628c2367 (44) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 (55) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: CSeq: 106 BYE (13) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Content-Length: 0 (17) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: (0) [Mar 10 12:08:07] --- (9 headers 0 lines) --- [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #270 [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '53c067b5491f609931bc7a0965b73fd3@24.123.23.170' of Request 106: Match Not Found [Mar 10 12:08:07] Really destroying SIP dialog '53c067b5491f609931bc7a0965b73fd3@24.123.23.170' Method: BYE [Mar 10 12:08:07] <--- 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=z9hG4bKve765c9620c30adb0386fbe1fadcc8616 CSeq: 39012 REGISTER Call-ID: 55ae66476793@24.123.23.170 From: ;tag=6726509f7fef65bf50359d83fd756f9f 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="23304027", algorithm=MD5, uri="sip:24.123.23.170", username="525", response="b0d75dbb247e3827e8c2fb498ab2edc3" <-------------> [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bKve765c9620c30adb0386fbe1fadcc8616 (82) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 39012 REGISTER (20) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Call-ID: 55ae66476793@24.123.23.170 (35) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=6726509f7fef65bf50359d83fd756f9f (66) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="23304027", algorithm=MD5, uri="sip:24.123.23.170", username="525", response="b0d75dbb247e3827e8c2fb498ab2edc3" (167) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:08:07] --- (13 headers 0 lines) --- [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 39012, ours 39012) [Mar 10 12:08:07] Using latest REGISTER request as basis request [Mar 10 12:08:07] Sending to 192.168.1.76 : 5060 (no NAT) [Mar 10 12:08:07] <--- 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=z9hG4bKve765c9620c30adb0386fbe1fadcc8616;received=192.168.1.76 From: ;tag=6726509f7fef65bf50359d83fd756f9f To: Call-ID: 55ae66476793@24.123.23.170 CSeq: 39012 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:08:07] NOTICE[18555]: chan_sip.c:8135 check_auth: Correct auth, but based on stale nonce received from '' [Mar 10 12:08:07] <--- 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=z9hG4bKve765c9620c30adb0386fbe1fadcc8616;received=192.168.1.76 From: ;tag=6726509f7fef65bf50359d83fd756f9f To: ;tag=as182bf0b1 Call-ID: 55ae66476793@24.123.23.170 CSeq: 39012 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3baabafb", stale=true Content-Length: 0 <------------> [Mar 10 12:08:07] Scheduling destruction of SIP dialog '55ae66476793@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:08:07] <--- 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=z9hG4bKpe1c089c552f0fcea2227e46bd5cd0765 CSeq: 43044 REGISTER Call-ID: 55ae664696c0@24.123.23.170 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 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="00273f8d", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="adf9acf187d165830272bcbd77f67cb3" <-------------> [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.98:5060;branch=z9hG4bKpe1c089c552f0fcea2227e46bd5cd0765 (82) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 43044 REGISTER (20) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Call-ID: 55ae664696c0@24.123.23.170 (35) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 (66) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="00273f8d", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="adf9acf187d165830272bcbd77f67cb3" (167) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:08:07] --- (13 headers 0 lines) --- [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 43044, ours 43044) [Mar 10 12:08:07] Using latest REGISTER request as basis request [Mar 10 12:08:07] Sending to 192.168.1.98 : 5060 (no NAT) [Mar 10 12:08:07] <--- 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=z9hG4bKpe1c089c552f0fcea2227e46bd5cd0765;received=192.168.1.98 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 To: Call-ID: 55ae664696c0@24.123.23.170 CSeq: 43044 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:08:07] NOTICE[18555]: chan_sip.c:8135 check_auth: Correct auth, but based on stale nonce received from '' [Mar 10 12:08:07] <--- 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=z9hG4bKpe1c089c552f0fcea2227e46bd5cd0765;received=192.168.1.98 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 To: ;tag=as08f52ac6 Call-ID: 55ae664696c0@24.123.23.170 CSeq: 43044 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5ea69cea", stale=true Content-Length: 0 <------------> [Mar 10 12:08:07] Scheduling destruction of SIP dialog '55ae664696c0@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:08:07] <--- SIP read from 192.168.1.85:5060 ---> BYE sip:593@24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKi2034d1cd4bddbaf9eab46925ade753a8 Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 388022 BYE From: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec To: ;tag=as628c2367 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 <-------------> [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: BYE sip:593@24.123.23.170 SIP/2.0 (33) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKi2034d1cd4bddbaf9eab46925ade753a8 (82) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 (55) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: CSeq: 388022 BYE (16) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec (74) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: ;tag=as628c2367 (42) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Max-Forwards: 70 (16) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Content-Length: 0 (17) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: (0) [Mar 10 12:08:07] --- (11 headers 0 lines) --- [Mar 10 12:08:07] <--- Transmitting (no NAT) to 192.168.1.85:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKi2034d1cd4bddbaf9eab46925ade753a8;received=192.168.1.85 From: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec To: ;tag=as628c2367 Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 388022 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:14774 sipsock_read: Invalid SIP message - rejected , no callid, len 508 [Mar 10 12:08:07] <--- 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=z9hG4bKve765c9620c30adb0386fbe1fadcc8616 CSeq: 39012 REGISTER Call-ID: 55ae66476793@24.123.23.170 From: ;tag=6726509f7fef65bf50359d83fd756f9f 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="23304027", algorithm=MD5, uri="sip:24.123.23.170", username="525", response="b0d75dbb247e3827e8c2fb498ab2edc3" <-------------> [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bKve765c9620c30adb0386fbe1fadcc8616 (82) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 39012 REGISTER (20) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Call-ID: 55ae66476793@24.123.23.170 (35) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=6726509f7fef65bf50359d83fd756f9f (66) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="23304027", algorithm=MD5, uri="sip:24.123.23.170", username="525", response="b0d75dbb247e3827e8c2fb498ab2edc3" (167) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:08:07] --- (13 headers 0 lines) --- [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 39012, ours 39012) [Mar 10 12:08:07] Using latest REGISTER request as basis request [Mar 10 12:08:07] Sending to 192.168.1.76 : 5060 (no NAT) [Mar 10 12:08:07] <--- 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=z9hG4bKve765c9620c30adb0386fbe1fadcc8616;received=192.168.1.76 From: ;tag=6726509f7fef65bf50359d83fd756f9f To: Call-ID: 55ae66476793@24.123.23.170 CSeq: 39012 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:08:07] NOTICE[18555]: chan_sip.c:8135 check_auth: Correct auth, but based on stale nonce received from '' [Mar 10 12:08:07] <--- 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=z9hG4bKve765c9620c30adb0386fbe1fadcc8616;received=192.168.1.76 From: ;tag=6726509f7fef65bf50359d83fd756f9f To: ;tag=as182bf0b1 Call-ID: 55ae66476793@24.123.23.170 CSeq: 39012 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2b76daeb", stale=true Content-Length: 0 <------------> [Mar 10 12:08:07] Scheduling destruction of SIP dialog '55ae66476793@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:08:07] <--- 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=z9hG4bKpe1c089c552f0fcea2227e46bd5cd0765 CSeq: 43044 REGISTER Call-ID: 55ae664696c0@24.123.23.170 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 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="00273f8d", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="adf9acf187d165830272bcbd77f67cb3" <-------------> [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.98:5060;branch=z9hG4bKpe1c089c552f0fcea2227e46bd5cd0765 (82) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 43044 REGISTER (20) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Call-ID: 55ae664696c0@24.123.23.170 (35) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 (66) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="00273f8d", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="adf9acf187d165830272bcbd77f67cb3" (167) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:08:07] --- (13 headers 0 lines) --- [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 43044, ours 43044) [Mar 10 12:08:07] Using latest REGISTER request as basis request [Mar 10 12:08:07] Sending to 192.168.1.98 : 5060 (no NAT) [Mar 10 12:08:07] <--- 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=z9hG4bKpe1c089c552f0fcea2227e46bd5cd0765;received=192.168.1.98 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 To: Call-ID: 55ae664696c0@24.123.23.170 CSeq: 43044 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:08:07] NOTICE[18555]: chan_sip.c:8135 check_auth: Correct auth, but based on stale nonce received from '' [Mar 10 12:08:07] <--- 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=z9hG4bKpe1c089c552f0fcea2227e46bd5cd0765;received=192.168.1.98 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 To: ;tag=as08f52ac6 Call-ID: 55ae664696c0@24.123.23.170 CSeq: 43044 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2ecca5f4", stale=true Content-Length: 0 <------------> [Mar 10 12:08:07] Scheduling destruction of SIP dialog '55ae664696c0@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:08:07] <--- 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=z9hG4bKk9ce3370c8525e6b0e8c0a1cb0bd7ae04 CSeq: 48253 REGISTER Call-ID: 55ae66469f7f@24.123.23.170 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 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="37147f29", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="9ea7effb31c722bbcec8b04fb74a36e1" <-------------> [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKk9ce3370c8525e6b0e8c0a1cb0bd7ae04 (82) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 48253 REGISTER (20) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Call-ID: 55ae66469f7f@24.123.23.170 (35) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 (66) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="37147f29", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="9ea7effb31c722bbcec8b04fb74a36e1" (167) [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:08:07] --- (13 headers 0 lines) --- [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:08:07] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 48253, ours 48253) [Mar 10 12:08:07] Using latest REGISTER request as basis request [Mar 10 12:08:07] Sending to 192.168.1.93 : 5060 (no NAT) [Mar 10 12:08:07] <--- 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=z9hG4bKk9ce3370c8525e6b0e8c0a1cb0bd7ae04;received=192.168.1.93 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 To: Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 48253 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:08:07] NOTICE[18555]: chan_sip.c:8135 check_auth: Correct auth, but based on stale nonce received from '' [Mar 10 12:08:07] <--- 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=z9hG4bKk9ce3370c8525e6b0e8c0a1cb0bd7ae04;received=192.168.1.93 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 To: ;tag=as549fb064 Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 48253 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="40989456", stale=true Content-Length: 0 <------------> [Mar 10 12:08:12] Scheduling destruction of SIP dialog '55ae66469f7f@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:08:09] DEBUG[18544]: devicestate.c:303 __ast_device_state_changed_literal: nNotification of state change to be queued on device/channel IAX2/616 ow [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:2008 __sip_autodestruct: Auto destroying SIP dialog '55ae66469f11@24.123.23.170' [Mar 10 12:08:12] Beginning asterisk shutdown.... [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:3107 sip_destroy: Destroying SIP dialog 55ae66469f11@24.123.23.170 [Mar 10 12:08:12] Really destroying SIP dialog '55ae66469f11@24.123.23.170' Method: REGISTER [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:2008 __sip_autodestruct: Auto destroying SIP dialog '55ae6647666e@24.123.23.170' [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:3107 sip_destroy: Destroying SIP dialog 55ae6647666e@24.123.23.170 [Mar 10 12:08:12] Really destroying SIP dialog '55ae6647666e@24.123.23.170' Method: REGISTER [Mar 10 12:08:12] NOTICE[18555]: chan_sip.c:7128 sip_reregister: -- Re-registration for 3173241052@sip.broadvoice.com [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4308 sip_alloc: Allocating new SIP dialog for 2c01f4fc3cc0317e799f35d0601622e1@24.123.23.170 - REGISTER (No RTP) [Mar 10 12:08:12] DEBUG[18527]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for IAX2 - 616 [Mar 10 12:08:12] DEBUG[18527]: chan_iax2.c:9636 iax2_devicestate: Checking device state for device 616 [Mar 10 12:08:12] DEBUG[18527]: chan_iax2.c:9644 iax2_devicestate: iax2_devicestate: Found peer. What's device state of 616? addr=1159759178, defaddr=0 maxms=0, lastms=0 [Mar 10 12:08:12] DEBUG[18527]: devicestate.c:287 do_state_change: Changing state for IAX2/616 - state 1 (Not in use) [Mar 10 12:08:12] DEBUG[18546]: chan_iax2.c:6538 socket_process: Packet arrived out of order (expecting 1, got 0) (frametype = 6, subclass = 13) [Mar 10 12:08:12] DEBUG[18546]: chan_iax2.c:6545 socket_process: Acking anyway [Mar 10 12:08:12] DEBUG[18629]: 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. [Mar 10 12:08:12] Executing last minute cleanups [Mar 10 12:08:12] == Destroying musiconhold processes [Mar 10 12:08:12] DEBUG[18522]: res_musiconhold.c:1125 ast_moh_destroy: killing 18529! [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:7287 transmit_register: Scheduled a registration timeout for sip.broadvoice.com id #286 [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:7334 transmit_register: >>> Re-using Auth data for 3173241052@sip.broadvoice.com [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:sip.broadvoice.com SIP/2.0 (39) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK66e022c4;rport (64) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: From: ;tag=as496cf9c0 (56) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: To: (39) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 2c01f4fc3cc0317e799f35d0601622e1@24.123.23.170 (55) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: CSeq: 107 REGISTER (18) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Max-Forwards: 70 (16) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Authorization: Digest username="3173241052", realm="BroadWorks", algorithm=MD5, uri="sip:sip.broadvoice.com", nonce="BroadWorksXez4b7xcwTti1x0kBW", response="4c77cc0c0f7db018ee91aff29607f2d8", opaque="", qop=auth, cnonce="17ab3b64", nc=00000005 (244) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Expires: 120 (12) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Contact: (39) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Event: registration (19) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Content-Length: 0 (17) [Mar 10 12:08:12] REGISTER 13 headers, 0 lines [Mar 10 12:08:12] REGISTER attempt 1 to 3173241052@sip.broadvoice.com [Mar 10 12:08:12] 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=z9hG4bK66e022c4;rport From: ;tag=as496cf9c0 To: Call-ID: 2c01f4fc3cc0317e799f35d0601622e1@24.123.23.170 CSeq: 107 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="3173241052", realm="BroadWorks", algorithm=MD5, uri="sip:sip.broadvoice.com", nonce="BroadWorksXez4b7xcwTti1x0kBW", response="4c77cc0c0f7db018ee91aff29607f2d8", opaque="", qop=auth, cnonce="17ab3b64", nc=00000005 Expires: 120 Contact: Event: registration Content-Length: 0 --- [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #287 [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:2008 __sip_autodestruct: Auto destroying SIP dialog '000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165' [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:3107 sip_destroy: Destroying SIP dialog 000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165 [Mar 10 12:08:12] Really destroying SIP dialog '000ff78d-ebb20006-635e2b11-3dc44fa0@192.168.1.165' Method: REGISTER [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:2008 __sip_autodestruct: Auto destroying SIP dialog '000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165' [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:3107 sip_destroy: Destroying SIP dialog 000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165 [Mar 10 12:08:12] Really destroying SIP dialog '000ff78d-ebb20005-5474240c-7cdf7903@192.168.1.165' Method: REGISTER [Mar 10 12:08:12] <--- SIP read from 192.168.1.85:5060 ---> SIP/2.0 200 OK To: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5bdd0182 From: ;tag=as628c2367 Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 106 BYE Content-Length: 0 Supported: sip-cc, sip-cc-01, replaces, timer Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK <-------------> [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: SIP/2.0 200 OK (14) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: To: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec (72) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5bdd0182 (58) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: From: ;tag=as628c2367 (44) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 (55) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: CSeq: 106 BYE (13) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Content-Length: 0 (17) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: (0) [Mar 10 12:08:12] --- (9 headers 0 lines) --- [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:14774 sipsock_read: Invalid SIP message - rejected , no callid, len 400 [Mar 10 12:08:12] <--- SIP read from 192.168.1.85:5060 ---> SIP/2.0 200 OK To: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5bdd0182 From: ;tag=as628c2367 Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 106 BYE Content-Length: 0 Supported: sip-cc, sip-cc-01, replaces, timer Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK <-------------> [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: SIP/2.0 200 OK (14) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: To: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec (72) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK5bdd0182 (58) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: From: ;tag=as628c2367 (44) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 (55) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: CSeq: 106 BYE (13) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Content-Length: 0 (17) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: (0) [Mar 10 12:08:12] --- (9 headers 0 lines) --- [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:14774 sipsock_read: Invalid SIP message - rejected , no callid, len 400 [Mar 10 12:08:12] <--- 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=z9hG4bKr99145258f1120ee55eb7cf537ae40442 CSeq: 48254 REGISTER Call-ID: 55ae66469f7f@24.123.23.170 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 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="4579536e", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="2ec967453cf838e40d8a27ed5d501d5f" <-------------> [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKr99145258f1120ee55eb7cf537ae40442 (82) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 48254 REGISTER (20) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Call-ID: 55ae66469f7f@24.123.23.170 (35) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 (66) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="4579536e", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="2ec967453cf838e40d8a27ed5d501d5f" (167) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:08:12] --- (13 headers 0 lines) --- [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:08:12] Using latest REGISTER request as basis request [Mar 10 12:08:12] Sending to 192.168.1.93 : 5060 (no NAT) [Mar 10 12:08:12] <--- 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=z9hG4bKr99145258f1120ee55eb7cf537ae40442;received=192.168.1.93 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 To: Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 48254 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:08:12] NOTICE[18555]: chan_sip.c:8135 check_auth: Correct auth, but based on stale nonce received from '' [Mar 10 12:08:12] <--- 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=z9hG4bKr99145258f1120ee55eb7cf537ae40442;received=192.168.1.93 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 To: ;tag=as549fb064 Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 48254 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="19da5213", stale=true Content-Length: 0 <------------> [Mar 10 12:08:12] Scheduling destruction of SIP dialog '55ae66469f7f@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:08:12] <--- 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=z9hG4bKr99145258f1120ee55eb7cf537ae40442 CSeq: 48254 REGISTER Call-ID: 55ae66469f7f@24.123.23.170 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 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="4579536e", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="2ec967453cf838e40d8a27ed5d501d5f" <-------------> [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKr99145258f1120ee55eb7cf537ae40442 (82) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 48254 REGISTER (20) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Call-ID: 55ae66469f7f@24.123.23.170 (35) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 (66) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="4579536e", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="2ec967453cf838e40d8a27ed5d501d5f" (167) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:08:12] --- (13 headers 0 lines) --- [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 48254, ours 48254) [Mar 10 12:08:12] Using latest REGISTER request as basis request [Mar 10 12:08:12] Sending to 192.168.1.93 : 5060 (no NAT) [Mar 10 12:08:12] <--- 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=z9hG4bKr99145258f1120ee55eb7cf537ae40442;received=192.168.1.93 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 To: Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 48254 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:08:12] NOTICE[18555]: chan_sip.c:8135 check_auth: Correct auth, but based on stale nonce received from '' [Mar 10 12:08:12] <--- 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=z9hG4bKr99145258f1120ee55eb7cf537ae40442;received=192.168.1.93 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 To: ;tag=as549fb064 Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 48254 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="03369f5d", stale=true Content-Length: 0 <------------> [Mar 10 12:08:12] Scheduling destruction of SIP dialog '55ae66469f7f@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:08:12] <--- SIP read from 192.168.1.85:5060 ---> BYE sip:593@24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKi2034d1cd4bddbaf9eab46925ade753a8 Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 388022 BYE From: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec To: ;tag=as628c2367 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 <-------------> [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: BYE sip:593@24.123.23.170 SIP/2.0 (33) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKi2034d1cd4bddbaf9eab46925ade753a8 (82) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 (55) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: CSeq: 388022 BYE (16) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec (74) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: ;tag=as628c2367 (42) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: Max-Forwards: 70 (16) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Content-Length: 0 (17) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: (0) [Mar 10 12:08:12] --- (11 headers 0 lines) --- [Mar 10 12:08:12] <--- Transmitting (no NAT) to 192.168.1.85:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bKi2034d1cd4bddbaf9eab46925ade753a8;received=192.168.1.85 From: 523 ;tag=a0b2f919869f2cd75f7879cc9d8200ec To: ;tag=as628c2367 Call-ID: 53c067b5491f609931bc7a0965b73fd3@24.123.23.170 CSeq: 388022 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:14774 sipsock_read: Invalid SIP message - rejected , no callid, len 508 [Mar 10 12:08:12] <--- 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=z9hG4bKve765c9620c30adb0386fbe1fadcc8616 CSeq: 39012 REGISTER Call-ID: 55ae66476793@24.123.23.170 From: ;tag=6726509f7fef65bf50359d83fd756f9f 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="23304027", algorithm=MD5, uri="sip:24.123.23.170", username="525", response="b0d75dbb247e3827e8c2fb498ab2edc3" <-------------> [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bKve765c9620c30adb0386fbe1fadcc8616 (82) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 39012 REGISTER (20) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Call-ID: 55ae66476793@24.123.23.170 (35) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=6726509f7fef65bf50359d83fd756f9f (66) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="23304027", algorithm=MD5, uri="sip:24.123.23.170", username="525", response="b0d75dbb247e3827e8c2fb498ab2edc3" (167) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:08:12] --- (13 headers 0 lines) --- [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:08:12] DEBUG[18522]: res_musiconhold.c:1139 ast_moh_destroy: mpg123 pid 18529 and child died after 19456 bytes read [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 39012, ours 39012) [Mar 10 12:08:12] Using latest REGISTER request as basis request [Mar 10 12:08:12] Sending to 192.168.1.76 : 5060 (no NAT) [Mar 10 12:08:12] Asterisk cleanly ending (0). [Mar 10 12:08:12] DEBUG[18522]: asterisk.c:1276 quit_handler: Asterisk ending (0). [Mar 10 12:08:12] <--- 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=z9hG4bKve765c9620c30adb0386fbe1fadcc8616;received=192.168.1.76 From: ;tag=6726509f7fef65bf50359d83fd756f9f To: Call-ID: 55ae66476793@24.123.23.170 CSeq: 39012 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:08:12] NOTICE[18555]: chan_sip.c:8135 check_auth: Correct auth, but based on stale nonce received from '' [Mar 10 12:08:12] <--- 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=z9hG4bKve765c9620c30adb0386fbe1fadcc8616;received=192.168.1.76 From: ;tag=6726509f7fef65bf50359d83fd756f9f To: ;tag=as182bf0b1 Call-ID: 55ae66476793@24.123.23.170 CSeq: 39012 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3af6ce45", stale=true Content-Length: 0 <------------> [Mar 10 12:08:12] Scheduling destruction of SIP dialog '55ae66476793@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:08:12] <--- 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=z9hG4bKpe1c089c552f0fcea2227e46bd5cd0765 CSeq: 43044 REGISTER Call-ID: 55ae664696c0@24.123.23.170 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 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="00273f8d", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="adf9acf187d165830272bcbd77f67cb3" <-------------> [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.98:5060;branch=z9hG4bKpe1c089c552f0fcea2227e46bd5cd0765 (82) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 43044 REGISTER (20) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Call-ID: 55ae664696c0@24.123.23.170 (35) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 (66) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="00273f8d", algorithm=MD5, uri="sip:24.123.23.170", username="529", response="adf9acf187d165830272bcbd77f67cb3" (167) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:08:12] --- (13 headers 0 lines) --- [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 43044, ours 43044) [Mar 10 12:08:12] Using latest REGISTER request as basis request [Mar 10 12:08:12] Sending to 192.168.1.98 : 5060 (no NAT) [Mar 10 12:08:12] <--- 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=z9hG4bKpe1c089c552f0fcea2227e46bd5cd0765;received=192.168.1.98 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 To: Call-ID: 55ae664696c0@24.123.23.170 CSeq: 43044 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:08:12] NOTICE[18555]: chan_sip.c:8135 check_auth: Correct auth, but based on stale nonce received from '' [Mar 10 12:08:12] <--- 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=z9hG4bKpe1c089c552f0fcea2227e46bd5cd0765;received=192.168.1.98 From: ;tag=40f10a0af6809cca23a4f7a5be3dec79 To: ;tag=as08f52ac6 Call-ID: 55ae664696c0@24.123.23.170 CSeq: 43044 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1a643e94", stale=true Content-Length: 0 <------------> [Mar 10 12:08:12] Scheduling destruction of SIP dialog '55ae664696c0@24.123.23.170' in 32000 ms (Method: REGISTER) [Mar 10 12:08:12] <--- 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=z9hG4bKr99145258f1120ee55eb7cf537ae40442 CSeq: 48254 REGISTER Call-ID: 55ae66469f7f@24.123.23.170 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 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="4579536e", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="2ec967453cf838e40d8a27ed5d501d5f" <-------------> [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 0: REGISTER sip:24.123.23.170 SIP/2.0 (34) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKr99145258f1120ee55eb7cf537ae40442 (82) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 2: CSeq: 48254 REGISTER (20) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 3: Call-ID: 55ae66469f7f@24.123.23.170 (35) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 4: From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 (66) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 5: To: (27) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 6: Contact: ;expires=60;q=0.50 (54) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 7: User-Agent: Uniden SIP Phone p2 Ver BS4.77 (42) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 9: Content-Length: 0 (17) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 10: Supported: sip-cc, sip-cc-01, replaces, timer (45) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK (62) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 12: Authorization: Digest realm="asterisk", nonce="4579536e", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="2ec967453cf838e40d8a27ed5d501d5f" (167) [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:4571 parse_request: Header 13: (0) [Mar 10 12:08:12] --- (13 headers 0 lines) --- [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:14590 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 10 12:08:12] DEBUG[18555]: chan_sip.c:14609 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 48254, ours 48254) [Mar 10 12:08:12] Using latest REGISTER request as basis request [Mar 10 12:08:12] Sending to 192.168.1.93 : 5060 (no NAT) [Mar 10 12:08:12] <--- 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=z9hG4bKr99145258f1120ee55eb7cf537ae40442;received=192.168.1.93 From: ;tag=ddb4f5df7dced8dad6a022bb1ca3e0a0 To: Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 48254 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 10 12:08:12] NOTICE[18555]: chan_sip.c:8135 check_auth: Correct auth, but based on stale nonce received from ''