Asterisk SVN-trunk-r83213M, Copyright (C) 1999 - 2007 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= NOTE: This is a development version of Asterisk, and should not be used in production installations. == Parsing '/etc/asterisk/asterisk.conf': == Found  == Parsing '/etc/asterisk/extconfig.conf':  == Found  == Binding sippeers to mysql/asterisk/sip_users  == Binding sipusers to mysql/asterisk/sip_users  == Binding voicemail to mysql/asterisk/voicemail_users  == Parsing '/etc/asterisk/logger.conf':  == Found Asterisk Event Logger Started /var/log/asterisk/event_log  Asterisk Dynamic Loader Starting:  == Parsing '/etc/asterisk/modules.conf':  == Found  == Parsing '/etc/asterisk/dnsmgr.conf':  == Found  == Parsing '/etc/asterisk/http.conf':  == Found  == Manager registered action Ping  == Manager registered action Events  == Manager registered action Logoff  == Manager registered action Login  == Manager registered action Challenge  == Manager registered action Hangup  == Manager registered action Status  == Manager registered action Setvar  == Manager registered action Getvar  == Manager registered action GetConfig  == Manager registered action GetConfigJSON  == Manager registered action UpdateConfig  == Manager registered action Redirect  == Manager registered action Originate  == Manager registered action Command  == Manager registered action ExtensionState  == Manager registered action AbsoluteTimeout  == Manager registered action MailboxStatus  == Manager registered action MailboxCount  == Manager registered action ListCommands  == Manager registered action SendText  == Manager registered action UserEvent  == Manager registered action WaitEvent  == Manager registered action CoreSettings  == Manager registered action CoreStatus  == Parsing '/etc/asterisk/manager.conf':  == Found  == Parsing '/etc/asterisk/cdr.conf':  == Found [Sep 20 15:51:00] NOTICE[8200]: cdr.c:1356 do_reload: CDR simple logging enabled.  == Parsing '/etc/asterisk/rtp.conf':  == Found  == RTP Allocating from port range 1024 -> 65535  == Parsing '/etc/asterisk/udptl.conf':  == Found  == UDPTL allocating from port range 4000 -> 4999  Asterisk PBX Core Initializing  Registering builtin applications:  [Answer]  == Registered application 'Answer'  [BackGround]  == Registered application 'BackGround'  [Busy]  == Registered application 'Busy'  [Congestion]  == Registered application 'Congestion'  [ExecIfTime]  == Registered application 'ExecIfTime'  [Goto]  == Registered application 'Goto'  [GotoIf]  == Registered application 'GotoIf'  [GotoIfTime]  == Registered application 'GotoIfTime'  [ImportVar]  == Registered application 'ImportVar'  [Hangup]  == Registered application 'Hangup'  [NoOp]  == Registered application 'NoOp'  [Progress]  == Registered application 'Progress'  [ResetCDR]  == Registered application 'ResetCDR'  [Ringing]  == Registered application 'Ringing'  [SayAlpha]  == Registered application 'SayAlpha'  [SayDigits]  == Registered application 'SayDigits'  [SayNumber]  == Registered application 'SayNumber'  [SayPhonetic]  == Registered application 'SayPhonetic'  [Set]  == Registered application 'Set'  [SetAMAFlags]  == Registered application 'SetAMAFlags'  [Wait]  == Registered application 'Wait'  [WaitExten]  == Registered application 'WaitExten'  [KeepAlive]  == Registered application 'KeepAlive'  == Manager registered action ShowDialPlan  == Manager registered action DBGet  == Manager registered action DBPut  == Manager registered action DBDel  == Manager registered action DBDelTree  == Parsing '/etc/asterisk/enum.conf':  == Found  Asterisk Dynamic Loader Starting:  == Parsing '/etc/asterisk/modules.conf':  == Found [Sep 20 15:51:00] NOTICE[8200]: loader.c:831 load_modules: 158 modules will be loaded.  == Registered application 'Bridge'  == Parsing '/etc/asterisk/features.conf':  == Found  -- Registered extension context 'parkedcalls'  -- Added extension '7000' priority 1 to parkedcalls  == Registered application 'ParkedCall'  == Registered application 'Park'  == Manager registered action ParkedCalls  == Manager registered action Park  == Manager registered action Bridge  res_features.so => (Call Features Resource)  == Parsing '/etc/asterisk/indications.conf':  == Found  -- Registered indication country 'at'  -- Registered indication country 'au'  -- Registered indication country 'bg'  -- Registered indication country 'br'  -- Registered indication country 'be'  -- Registered indication country 'ch'  -- Registered indication country 'cl'  -- Registered indication country 'cn'  -- Registered indication country 'cz'  -- Registered indication country 'de'  -- Registered indication country 'dk'  -- Registered indication country 'ee'  -- Registered indication country 'es'  -- Registered indication country 'fi'  -- Registered indication country 'fr'  -- Registered indication country 'gr'  -- Registered indication country 'hu'  -- Registered indication country 'il'  -- Registered indication country 'in'  -- Registered indication country 'it'  -- Registered indication country 'lt'  -- Registered indication country 'jp'  -- Registered indication country 'mx'  -- Registered indication country 'my'  -- Registered indication country 'nl'  -- Registered indication country 'no'  -- Registered indication country 'nz'  -- Registered indication country 'pl'  -- Registered indication country 'pt'  -- Registered indication country 'ru'  -- Registered indication country 'se'  -- Registered indication country 'sg'  -- Registered indication country 'th'  -- Registered indication country 'uk'  -- Registered indication country 'us'  -- Registered indication country 'us-o'  -- Registered indication country 'tw'  -- Registered indication country 've'  -- Registered indication country 'za'  -- Setting default indication country to 'il'  == Registered application 'PlayTones'  == Registered application 'StopPlayTones'  res_indications.so => (Region-specific tones)  == Parsing '/etc/asterisk/smdi.conf':  == Found [Sep 20 15:51:00] WARNING[8200]: res_smdi.c:715 load_module: No SMDI interfaces are available to listen on, not starting SDMI listener.  == Registered application 'Monitor'  == Registered application 'StopMonitor'  == Registered application 'ChangeMonitor'  == Registered application 'PauseMonitor'  == Registered application 'UnpauseMonitor'  == Manager registered action Monitor  == Manager registered action StopMonitor  == Manager registered action ChangeMonitor  == Manager registered action PauseMonitor  == Manager registered action UnpauseMonitor  res_monitor.so => (Call Monitoring Resource)  == AGI Command 'answer' registered  == AGI Command 'channel status' registered  == AGI Command 'database del' registered  == AGI Command 'database deltree' registered  == AGI Command 'database get' registered  == AGI Command 'database put' registered  == AGI Command 'exec' registered  == AGI Command 'get data' registered  == AGI Command 'get full variable' registered  == AGI Command 'get option' registered  == AGI Command 'get variable' registered  == AGI Command 'hangup' registered  == AGI Command 'noop' registered  == AGI Command 'receive char' registered  == AGI Command 'receive text' registered  == AGI Command 'record file' registered  == AGI Command 'say alpha' registered  == AGI Command 'say digits' registered  == AGI Command 'say number' registered  == AGI Command 'say phonetic' registered  == AGI Command 'say date' registered  == AGI Command 'say time' registered  == AGI Command 'say datetime' registered  == AGI Command 'send image' registered  == AGI Command 'send text' registered  == AGI Command 'set autohangup' registered  == AGI Command 'set callerid' registered  == AGI Command 'set context' registered  == AGI Command 'set extension' registered  == AGI Command 'set music' registered  == AGI Command 'set priority' registered  == AGI Command 'set variable' registered  == AGI Command 'stream file' registered  == AGI Command 'control stream file' registered  == AGI Command 'tdd mode' registered  == AGI Command 'verbose' registered  == AGI Command 'wait for digit' registered  == Registered application 'DeadAGI'  == Registered application 'EAGI'  == Registered application 'AGI'  res_agi.so => (Asterisk Gateway Interface (AGI))  res_ael_share.so => (share-able code for AEL)  res_speech.so => (Generic Speech Recognition API)  == Registered application 'MusicOnHold'  == Registered application 'WaitMusicOnHold'  == Registered application 'SetMusicOnHold'  == Registered application 'StartMusicOnHold'  == Registered application 'StopMusicOnHold'  == Parsing '/etc/asterisk/musiconhold.conf':  == Found  res_musiconhold.so => (Music On Hold Resource)  == Parsing '/etc/asterisk/res_mysql.conf':  == Found [Sep 20 15:51:00] NOTICE[8200]: config.c:1766 ast_config_engine_register: Registered Config Engine mysql  == MySQL RealTime driver loaded.  res_config_mysql.so => (MySQL RealTime Configuration Driver)  == Registered application 'ControlPlayback'  app_controlplayback.so => (Control Playback Application)  == Registered application 'ADSIProg'  app_adsiprog.so => (Asterisk ADSI Programming Application)  pbx_spool.so => (Outgoing Spool Support)  == Registered channel type 'Local' (Local Proxy Channel Driver)  chan_local.so => (Local Proxy Channel)  == Registered application 'Dictate'  app_dictate.so => (Virtual Dictation Machine)  == Registered channel type 'Feature' (Feature Proxy Channel Driver)  chan_features.so => (Feature Proxy Channel)  == Registered file format iLBC, extension(s) ilbc  format_ilbc.so => (Raw iLBC data)  == Registered application 'Zapateller'  app_zapateller.so => (Block Telemarketers with Special Information Tone)  res_limit.so => (Resource limits)  == Registered application 'DISA'  app_disa.so => (DISA (Direct Inward System Access) Application) SIP channel loading...  == Parsing '/etc/asterisk/sip.conf':  == Found  == Parsing '/etc/asterisk/users.conf':  == Found  == SIP Listening on 0.0.0.0:5060  == Using TOS bits 0  == Using CoS mark 4  == Parsing '/etc/asterisk/sip_notify.conf':  == Found  == Registered channel type 'SIP' (Session Initiation Protocol (SIP))  == Registered application 'SIPDtmfMode'  == Registered application 'SIPAddHeader'  == Registered custom function SIP_HEADER  == Registered custom function SIPPEER  == Registered custom function SIPCHANINFO  == Registered custom function CHECKSIPDOMAIN  == Manager registered action SIPpeers  == Manager registered action SIPshowpeer  chan_sip.so => (Session Initiation Protocol (SIP))  == Registered translator 'ilbctolin' from format ilbc to slin, cost 5144  == Registered translator 'lintoilbc' from format slin to ilbc, cost 28820  codec_ilbc.so => (iLBC Coder/Decoder)  == Registered application 'NBScat'  app_nbscat.so => (Silly NBS Stream Application)  == Registered application 'TrySystem'  == Registered application 'System'  app_system.so => (Generic System() application)  == Registered application 'SendURL'  app_url.so => (Send URL Applications)  == Registered format 'jpg' (JPEG (Joint Picture Experts Group))  format_jpeg.so => (JPEG (Joint Picture Experts Group) Image Format)  == Registered file format h264, extension(s) h264  format_h264.so => (Raw H.264 data)  == Registered application 'Authenticate'  app_authenticate.so => (Authentication Application)  == Registered application 'SMS'  app_sms.so => (SMS/PSTN handler)  == Registered translator 'alawtoulaw' from format alaw to ulaw, cost 60  == Registered translator 'ulawtoalaw' from format ulaw to alaw, cost 58  codec_a_mu.so => (A-law and Mulaw direct Coder/Decoder)  == Registered custom function ENUMRESULT  == Registered custom function ENUMQUERY  == Registered custom function ENUMLOOKUP  == Registered custom function TXTCIDNAME  func_enum.so => (ENUM related dialplan functions)  == Registered application 'Page'  app_page.so => (Page Multiple Phones)  == Parsing '/etc/asterisk/cdr_manager.conf':  == Found  cdr_manager.so => (Asterisk Manager Interface CDR Backend)  == Registered custom function CURL  func_curl.so => (Load external URL)  == Registered file format g729, extension(s) g729  format_g729.so => (Raw G729 data)  == Parsing '/etc/asterisk/cdr.conf':  == Found  == Registered application 'Milliwatt'  app_milliwatt.so => (Digital Milliwatt (mu-law) Test Application)  == Registered application 'Exec'  == Registered application 'TryExec'  == Registered application 'ExecIf'  app_exec.so => (Executes dialplan applications)  == Parsing '/etc/asterisk/cdr_custom.conf':  == Found  cdr_custom.so => (Customizable Comma Separated Values CDR Backend)  pbx_loopback.so => (Loopback Switch)  == Registered file format ogg_vorbis, extension(s) ogg  format_ogg_vorbis.so => (OGG/Vorbis audio)  == Registered file format g723sf, extension(s) g723|g723sf  format_g723.so => (G.723.1 Simple Timestamp File Format)  == Parsing '/etc/asterisk/cdr_mysql.conf':  == Found  cdr_addon_mysql.so => (MySQL CDR Backend)  == Registered custom function URIDECODE  == Registered custom function URIENCODE  func_uri.so => (URI encode/decode dialplan functions)  == Registered application 'WaitForSilence'  app_waitforsilence.so => (Wait For Silence)  == Registered application 'ZapRAS'  app_zapras.so => (Zaptel ISDN Remote Access Server)  == Registered custom function IFMODULE  func_module.so => (Checks if Asterisk module is loaded in memory)  == Registered file format sln, extension(s) sln|raw  format_sln.so => (Raw Signed Linear Audio support (SLN))  == Parsing '/etc/asterisk/zapata.conf':  == Found  == Parsing '/etc/asterisk/users.conf':  == Found  -- Automatically generated pseudo channel  == Registered channel type 'Zap' (Zapata Telephony Driver)  == Manager registered action ZapTransfer  == Manager registered action ZapHangup  == Manager registered action ZapDialOffhook  == Manager registered action ZapDNDon  == Manager registered action ZapDNDoff  == Manager registered action ZapShowChannels  == Manager registered action ZapRestart  chan_zap.so => (Zapata Telephony)  == Parsing '/etc/asterisk/skinny.conf':  == Found  == Skinny listening on 0.0.0.0:2000  == Registered channel type 'Skinny' (Skinny Client Control Protocol (Skinny))  == Registered application 'While'  == Registered application 'EndWhile'  == Registered application 'ExitWhile'  == Registered application 'ContinueWhile'  app_while.so => (While Loops and Conditional Execution)  == Registered custom function TIMEOUT  func_timeout.so => (Channel timeout dialplan functions) [Sep 20 15:51:00] WARNING[8200]: cdr_sqlite3_custom.c:85 load_config: cdr_sqlite3_custom: Failed to load configuration file. Module not activated.  == Parsing '/etc/asterisk/amd.conf':  == Found  -- AMD defaults: initialSilence [2500] greeting [1500] afterGreetingSilence [800] totalAnalysisTime [5000] minimumWordLength [100] betweenWordsSilence [50] maximumNumberOfWords [3] silenceThreshold [256]  == Registered application 'AMD'  app_amd.so => (Answering Machine Detection Application)  == Registered custom function DB  == Registered custom function DB_EXISTS  == Registered custom function DB_DELETE  func_db.so => (Database (astdb) related dialplan functions)  == Registered file format h263, extension(s) h263  format_h263.so => (Raw H.263 data)  == Registered application 'MinivmRecord'  == Registered application 'MinivmGreet'  == Registered application 'MinivmNotify'  == Registered application 'MinivmDelete'  == Registered application 'MinivmAccMess'  == Registered custom function MINIVMACCOUNT  == Registered custom function MINIVMCOUNTER [Sep 20 15:51:00] WARNING[8200]: app_minivm.c:2394 load_config: Failed to load configuration file. Module activated with default settings.  app_minivm.so => (Mini VoiceMail (A minimal Voicemail e-mail System))  == Registered application 'ZapScan'  app_zapscan.so => (Scan Zap channels application)  == Registered application 'ChannelRedirect'  app_channelredirect.so => (Redirects a given channel to a dialplan target)  == Parsing '/etc/asterisk/adsi.conf':  == Found  res_adsi.so => (ADSI Resource)  == Registered application 'UserEvent'  app_userevent.so => (Custom User Event Application)  == Registered application 'SendText'  app_sendtext.so => (Send Text Applications)  == Registered file format mp3, extension(s) mp3  format_mp3.so => (MP3 format [Any rate but 8000hz mono is optimal])  == Registered custom function GLOBAL  func_global.so => (Global variable dialplan functions)  == Registered custom function CUT  == Registered custom function SORT  func_cut.so => (Cut out information from a string)  == Parsing '/etc/asterisk/codecs.conf':  == Found  -- codec_gsm: using generic PLC  == Registered translator 'gsmtolin' from format gsm to slin, cost 2016  == Registered translator 'lintogsm' from format slin to gsm, cost 4801  codec_gsm.so => (GSM Coder/Decoder)  == Parsing '/etc/asterisk/alarmreceiver.conf':  == Found  == Registered application 'AlarmReceiver'  app_alarmreceiver.so => (Alarm Receiver for Asterisk)  == Registered custom function CHANNEL  func_channel.so => (Channel information dialplan function)  == Registered application 'SendImage'  app_image.so => (Image Transmission Application)  == Registered custom function ISNULL  == Registered custom function SET  == Registered custom function EXISTS  == Registered custom function IF  == Registered custom function IFTIME  == Registered custom function IMPORT  func_logic.so => (Logical dialplan functions)  == Registered application 'SpeechCreate'  == Registered application 'SpeechLoadGrammar'  == Registered application 'SpeechUnloadGrammar'  == Registered application 'SpeechActivateGrammar'  == Registered application 'SpeechDeactivateGrammar'  == Registered application 'SpeechStart'  == Registered application 'SpeechBackground'  == Registered application 'SpeechDestroy'  == Registered application 'SpeechProcessingSound'  == Registered custom function SPEECH  == Registered custom function SPEECH_SCORE  == Registered custom function SPEECH_TEXT  == Registered custom function SPEECH_GRAMMAR  == Registered custom function SPEECH_ENGINE  == Registered custom function SPEECH_RESULTS_TYPE  app_speech_utils.so => (Dialplan Speech Applications)  == Registered application 'GetCPEID'  app_getcpeid.so => (Get ADSI CPE ID)  == Parsing '/etc/asterisk/voicemail.conf':  == Found  == Parsing '/etc/asterisk/users.conf':  == Found  == Registered application 'VoiceMail'  == Registered application 'VoiceMailMain'  == Registered application 'MailboxExists'  == Registered application 'VMAuthenticate'  == Registered custom function MAILBOX_EXISTS  == Manager registered action VoicemailUsersList  app_voicemail.so => (Comedian Mail (Voicemail System))  == Registered custom function IAXPEER  == Registered custom function IAXVAR [Sep 20 15:51:00] WARNING[8200]: chan_iax2.c:11265 load_module: Unable to open IAX timing interface: No such file or directory  == Registered application 'IAX2Provision'  == Manager registered action IAXpeers  == Manager registered action IAXnetstats  == Parsing '/etc/asterisk/iax.conf':  == Found  == Parsing '/etc/asterisk/users.conf':  == Found  == Using TOS bits 0  == Using CoS mark 0  == Binding IAX2 to default address 0.0.0.0:4569  == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2))  == 10 helper threads started  == IAX Ready and Listening  == Loaded firmware 'iaxy.bin'  == Parsing '/etc/asterisk/iaxprov.conf':  == Found  -- Loaded provisioning template 'default'  chan_iax2.so => (Inter Asterisk eXchange (Ver 2))  == Registered application 'MP3Player'  app_mp3.so => (Silly MP3 Application)  == Registered application 'SayCountPL'  app_saycountpl.so => (Say polish counting words)  == Registered application 'NoCDR'  app_cdr.so => (Tell Asterisk to not maintain a CDR for the current call)  -- Loaded PUBLIC key 'iaxtel'  -- Loaded PUBLIC key 'freeworlddialup'  res_crypto.so => (Cryptographic Digital Signatures)  == Parsing '/etc/asterisk/codecs.conf':  == Found  -- codec_ulaw: using generic PLC  == Registered translator 'ulawtolin' from format ulaw to slin, cost 83  == Registered translator 'lintoulaw' from format slin to ulaw, cost 64  codec_ulaw.so => (mu-Law Coder/Decoder)  == Registered application 'Pickup'  app_directed_pickup.so => (Directed Call Pickup Application) [Sep 20 15:51:00] NOTICE[8200]: pbx_ael.c:1680 pbx_load_module: Starting AEL load process. [Sep 20 15:51:00] NOTICE[8200]: pbx_ael.c:1687 pbx_load_module: AEL load process: calculated config file name '/etc/asterisk/extensions.ael'. [Sep 20 15:51:00] NOTICE[8200]: pbx_ael.c:1695 pbx_load_module: AEL load process: parsed config file name '/etc/asterisk/extensions.ael'. [Sep 20 15:51:00] WARNING[8200]: ael/pval.c:821 check_includes: Warning: file /etc/asterisk/extensions.ael, line 93-96: The included context 'huji-local-hints' cannot be found. [Sep 20 15:51:00] WARNING[8200]: ../main/ast_expr2.y:900 op_func: Hey! chan is NULL. [Sep 20 15:51:00] WARNING[8200]: ../main/ast_expr2.y:902 op_func: Hey! could not find func LEN. [Sep 20 15:51:00] ERROR[8200]: ../main/ast_expr2.y:920 op_func: Error! 'LEN' doesn't appear to be an available function![Sep 20 15:51:00] NOTICE[8200]: pbx_ael.c:1698 pbx_load_module: AEL load process: checked config file name '/etc/asterisk/extensions.ael'.  -- Registered extension context 'change_GL_caller_id'  -- Registered extension context 'outbound'  -- Registered extension context 'huji-local'  -- Including context 'parkedcalls' in context 'huji-local'  -- Including context 'huji-local-hints' in context 'huji-local'  -- Registered extension context 'huji-local-gr'  -- Including context 'huji-local' in context 'huji-local-gr'  -- Registered extension context 'huji-local-ms'  -- Including context 'huji-local' in context 'huji-local-ms'  -- Registered extension context 'huji-local-ek'  -- Including context 'huji-local' in context 'huji-local-ek'  -- Registered extension context 'huji-local-ag'  -- Including context 'huji-local' in context 'huji-local-ag'  -- Registered extension context 'huji-outbound-gr'  -- Including context 'outbound' in context 'huji-outbound-gr'  -- Registered extension context 'huji-outbound-ms'  -- Including context 'outbound' in context 'huji-outbound-ms'  -- Registered extension context 'huji-outbound-ek'  -- Including context 'outbound' in context 'huji-outbound-ek'  -- Registered extension context 'huji-outbound-ag'  -- Including context 'outbound' in context 'huji-outbound-ag'  -- Registered extension context 'huji-remote-gr'  -- Including context 'huji-local-gr' in context 'huji-remote-gr'  -- Including context 'huji-outbound-gr' in context 'huji-remote-gr'  -- Registered extension context 'huji-remote-ms'  -- Including context 'huji-local-ms' in context 'huji-remote-ms'  -- Including context 'huji-outbound-ms' in context 'huji-remote-ms'  -- Registered extension context 'huji-remote-ek'  -- Including context 'huji-local-ek' in context 'huji-remote-ek'  -- Including context 'huji-outbound-ek' in context 'huji-remote-ek'  -- Registered extension context 'huji-remote-ag'  -- Including context 'huji-local-ag' in context 'huji-remote-ag'  -- Including context 'huji-outbound-ag' in context 'huji-remote-ag'  -- Registered extension context 'default'  -- Including context 'huji-local-gr' in context 'default'  -- Registered extension context 'HUJI_IVR_MAIN_STUD'  -- Registered extension context 'HUJI_IVR_MAIN_TAHZUKANIT'  -- Added extension 's' priority 1 to change_GL_caller_id  -- Added extension 's' priority 2 to change_GL_caller_id  -- Added extension 's' priority 3 to change_GL_caller_id  -- Added extension 's' priority 4 to change_GL_caller_id  -- Added extension 's' priority 5 to change_GL_caller_id  -- Added extension 's' priority 6 to change_GL_caller_id  -- Added extension 's' priority 7 to change_GL_caller_id  -- Added extension 's' priority 8 to change_GL_caller_id  -- Added extension 's' priority 9 to change_GL_caller_id  -- Added extension 's' priority 10 to change_GL_caller_id  -- Added extension 's' priority 11 to change_GL_caller_id  -- Added extension 's' priority 12 to change_GL_caller_id  -- Added extension 's' priority 13 to change_GL_caller_id  -- Added extension 's' priority 14 to change_GL_caller_id  -- Added extension 's' priority 15 to change_GL_caller_id  -- Added extension '_0[23489]XXXXXXX' priority 1 to outbound  -- Added extension '_0[23489]XXXXXXX' priority 2 to outbound  -- Added extension '_0[23489]XXXXXXX' priority 3 to outbound  -- Added extension '_07XXXXXXXX' priority 1 to outbound  -- Added extension '_07XXXXXXXX' priority 2 to outbound  -- Added extension '_07XXXXXXXX' priority 3 to outbound  -- Added extension '_05XXXXXXXX' priority 1 to outbound  -- Added extension '_05XXXXXXXX' priority 2 to outbound  -- Added extension '_05XXXXXXXX' priority 3 to outbound  -- Added extension '_1800XXXXXX' priority 1 to outbound  -- Added extension '_1800XXXXXX' priority 2 to outbound  -- Added extension '_1700XXXXXX' priority 1 to outbound  -- Added extension '_1700XXXXXX' priority 2 to outbound  -- Added extension '_1700XXXXXX' priority 3 to outbound  -- Added extension '_1599XXXXXX' priority 1 to outbound  -- Added extension '_1599XXXXXX' priority 2 to outbound  -- Added extension '_1599XXXXXX' priority 3 to outbound  -- Added extension '_*XXXX' priority 1 to outbound  -- Added extension '_*XXXX' priority 2 to outbound  -- Added extension '_*XXXX' priority 3 to outbound  -- Added extension '_1[0123489]X' priority 1 to outbound  -- Added extension '_1[0123489]X' priority 2 to outbound  -- Added extension '_97207227056XX' priority 1 to huji-local  -- Added extension '_97207227056XX' priority 2 to huji-local  -- Added extension '_97207227056XX' priority 3 to huji-local  -- Added extension '_97207227056XX' priority 4 to huji-local  -- Added extension '_97207227056XX' priority 5 to huji-local  -- Added extension '_97207227056XX' priority 6 to huji-local  -- Added extension '_97207227056XX' priority 7 to huji-local  -- Added extension '_97207227056XX' priority 8 to huji-local  -- Added extension '_97207227056XX' priority 9 to huji-local  -- Added extension '_97207227056XX' priority 10 to huji-local  -- Added extension '_97207227056XX' priority 11 to huji-local  -- Added extension '_97207227056XX' priority 12 to huji-local  -- Added extension '_97207227056XX' priority 13 to huji-local  -- Added extension '_97207227056XX' priority 14 to huji-local  -- Added extension '_97207227056XX' priority 15 to huji-local  -- Added extension '_97207227056XX' priority 16 to huji-local  -- Added extension '_408XXXX' priority 1 to huji-local  -- Added extension '_408XXXX' priority 2 to huji-local  -- Added extension '_408XXXX' priority 3 to huji-local  -- Added extension '_408XXXX' priority 4 to huji-local  -- Added extension '_408XXXX' priority 5 to huji-local  -- Added extension '_408XXXX' priority 6 to huji-local  -- Added extension '_408XXXX' priority 7 to huji-local  -- Added extension '_408XXXX' priority 8 to huji-local  -- Added extension '_408XXXX' priority 9 to huji-local  -- Added extension '80666' priority 1 to huji-local  -- Added extension 'asterisk' priority 1 to huji-local  -- Added extension '80698' priority 1 to huji-local  -- Added extension '80697' priority 1 to huji-local  -- Added extension '_8069X' priority 1 to huji-local  -- Added extension '_8069X' priority 2 to huji-local  -- Added extension '_806XX' priority 1 to huji-local  -- Added extension '_806XX' priority 2 to huji-local  -- Added extension '_806XX' priority 3 to huji-local  -- Added extension '_806XX' priority 4 to huji-local  -- Added extension '_806XX' priority 5 to huji-local  -- Added extension '_806XX' priority 6 to huji-local  -- Added extension '_806XX' priority 7 to huji-local  -- Added extension '_806XX' priority 8 to huji-local  -- Added extension '_806XX' priority 9 to huji-local  -- Added extension '_806XX' priority 10 to huji-local  -- Added extension '_806XX' priority 11 to huji-local  -- Added extension '_806XX' priority 12 to huji-local  -- Added extension '_806XX' priority 13 to huji-local  -- Added extension '_806XX' priority 14 to huji-local  -- Added extension '_806XX' priority 15 to huji-local  -- Added extension '_806XX' priority 16 to huji-local  -- Added extension '_806XX' priority 17 to huji-local  -- Added extension '_806XX' priority 18 to huji-local  -- Added extension '_806XX' priority 19 to huji-local  -- Added extension '_806XX' priority 20 to huji-local  -- Added extension '_806XX' priority 21 to huji-local  -- Added extension '_806XX' priority 22 to huji-local  -- Added extension '_806XX' priority 23 to huji-local  -- Added extension '_806XX' priority 24 to huji-local  -- Added extension '_806XX' priority 25 to huji-local  -- Added extension '_sw-9-.' priority 10 to huji-local  -- Added extension '_sw-9-.' priority 11 to huji-local  -- Added extension 'sw-9-NOANSWER' priority 10 to huji-local  -- Added extension 'sw-9-NOANSWER' priority 11 to huji-local  -- Added extension 'sw-9-NOANSWER' priority 12 to huji-local  -- Added extension 'sw-9-NOANSWER' priority 13 to huji-local  -- Added extension 'sw-9-NOANSWER' priority 14 to huji-local  -- Added extension 'sw-9-NOANSWER' priority 15 to huji-local  -- Added extension 'sw-9-CONGESTION' priority 10 to huji-local  -- Added extension 'sw-9-CONGESTION' priority 11 to huji-local  -- Added extension 'sw-9-CONGESTION' priority 12 to huji-local  -- Added extension 'sw-9-CONGESTION' priority 13 to huji-local  -- Added extension 'sw-9-CONGESTION' priority 14 to huji-local  -- Added extension 'sw-9-CONGESTION' priority 15 to huji-local  -- Added extension 'sw-9-CHANUNAVAIL' priority 10 to huji-local  -- Added extension 'sw-9-BUSY' priority 10 to huji-local  -- Added extension 'h' priority 1 to huji-local  -- Added extension 'h' priority 2 to huji-local  -- Added extension 'h' priority 3 to huji-local  -- Added extension 'h' priority 4 to huji-local  -- Added extension 'h' priority 5 to huji-local  -- Added extension 'h' priority 6 to huji-local  -- Added extension 'h' priority 7 to huji-local  -- Added extension 'h' priority 8 to huji-local  -- Added extension 'h' priority 9 to huji-local  -- Added extension 'h' priority 10 to huji-local  -- Added extension 'h' priority 11 to huji-local  -- Added extension 'h' priority 12 to huji-local  -- Added extension 'h' priority 13 to huji-local  -- Added extension 'h' priority 14 to huji-local  -- Added extension 'h' priority 15 to huji-local  -- Added extension 'h' priority 16 to huji-local  -- Added extension 'h' priority 17 to huji-local  -- Added extension 'h' priority 18 to huji-local  -- Added extension 'h' priority 19 to huji-local  -- Added extension 'h' priority 20 to huji-local  -- Added extension '*42' priority 1 to huji-local  -- Added extension '*42' priority 2 to huji-local  -- Added extension '*42' priority 3 to huji-local  -- Added extension '*41' priority 1 to huji-local  -- Added extension '*41' priority 2 to huji-local  -- Added extension '*41' priority 3 to huji-local  -- Added extension '*41' priority 4 to huji-local  -- Added extension '_80[0-5]XX' priority 1 to huji-local  -- Added extension '_80[0-5]XX' priority 2 to huji-local  -- Added extension '_8[1-9]XXX' priority 1 to huji-local  -- Added extension '_8[1-9]XXX' priority 2 to huji-local  -- Added extension '_8[1-9]XXX' priority 3 to huji-local  -- Added extension '_8[1-9]XXX' priority 4 to huji-local  -- Added extension '_8[1-9]XXX' priority 5 to huji-local  -- Added extension '_8[1-9]XXX' priority 6 to huji-local  -- Added extension '_8[1-9]XXX' priority 7 to huji-local  -- Added extension '_8[1-9]XXX' priority 8 to huji-local  -- Added extension '_8[1-9]XXX' priority 9 to huji-local  -- Added extension '_8[1-9]XXX' priority 10 to huji-local  -- Added extension '_8[1-9]XXX' priority 11 to huji-local  -- Added extension '_8[1-9]XXX' priority 12 to huji-local  -- Added extension '_8[1-9]XXX' priority 13 to huji-local  -- Added extension '_8[1-9]XXX' priority 14 to huji-local  -- Added extension '_8[1-9]XXX' priority 15 to huji-local  -- Added extension '_8[1-9]XXX' priority 16 to huji-local  -- Added extension '_2XXXX' priority 1 to huji-local  -- Added extension '_2XXXX' priority 2 to huji-local  -- Added extension '_2XXXX' priority 3 to huji-local  -- Added extension '_2XXXX' priority 4 to huji-local  -- Added extension '_2XXXX' priority 5 to huji-local  -- Added extension '_2XXXX' priority 6 to huji-local  -- Added extension '_6XXXX' priority 1 to huji-local  -- Added extension '_6XXXX' priority 2 to huji-local  -- Added extension '7000' priority 1 to huji-local  -- Added extension '7000' priority 2 to huji-local  -- Added extension '_99990XXXXXXXX' priority 1 to huji-local  -- Added extension '_99990XXXXXXXX' priority 2 to huji-local  -- Added extension '_99990XXXXXXXX' priority 3 to huji-local  -- Added extension '99999' priority 1 to huji-local  -- Added extension '99999' priority 2 to huji-local  -- Added extension '99999' priority 3 to huji-local  -- Added extension '99999' priority 4 to huji-local  -- Added extension '99999' priority 5 to huji-local  -- Added extension '99999' priority 6 to huji-local  -- Added extension '99999' priority 7 to huji-local  -- Added extension '99999' priority 8 to huji-local  -- Added extension '99999' priority 9 to huji-local  -- Added extension '99999' priority 10 to huji-local  -- Added extension 'Start' priority 1 to HUJI_IVR_MAIN_STUD  -- Added extension 'Start' priority 2 to HUJI_IVR_MAIN_STUD  -- Added extension 'MainMenu' priority 1 to HUJI_IVR_MAIN_STUD  -- Added extension 'MainMenu' priority 2 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 1 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 2 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 3 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 4 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 5 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 6 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 7 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 8 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 9 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 10 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 11 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 12 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 13 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 14 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 15 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 16 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 17 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 18 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 19 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 20 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 21 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 22 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 23 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 24 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 25 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 26 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 27 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 28 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 29 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 30 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 31 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 32 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 33 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 34 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 35 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 36 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 37 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 38 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 39 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 40 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 41 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 42 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 43 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 44 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 45 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 46 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 47 to HUJI_IVR_MAIN_STUD  -- Added extension '1' priority 48 to HUJI_IVR_MAIN_STUD  -- Added extension '2' priority 1 to HUJI_IVR_MAIN_STUD  -- Added extension '2' priority 2 to HUJI_IVR_MAIN_STUD  -- Added extension '3' priority 1 to HUJI_IVR_MAIN_STUD  -- Added extension '3' priority 2 to HUJI_IVR_MAIN_STUD  -- Added extension '4' priority 1 to HUJI_IVR_MAIN_STUD  -- Added extension '4' priority 2 to HUJI_IVR_MAIN_STUD  -- Added extension 'Start' priority 1 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension 'Start' priority 2 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension 'Start' priority 3 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension 'Start' priority 4 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension 'Start' priority 5 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension 'Start' priority 6 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension 'Start' priority 7 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension 'Start' priority 8 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension 'Start' priority 9 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension 'Start' priority 10 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension 'Start' priority 11 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension 'Start' priority 12 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension 'Start' priority 13 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension 'Start' priority 14 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension 'Start' priority 15 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension 'Start' priority 16 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension '1' priority 1 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension '1' priority 2 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension '1' priority 3 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension '1' priority 4 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension '1' priority 5 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension '1' priority 6 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension '1' priority 7 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension '2' priority 1 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension '2' priority 2 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension '2' priority 3 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension '2' priority 4 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension '2' priority 5 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension '2' priority 6 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension '2' priority 7 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension '3' priority 1 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension '3' priority 2 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension '3' priority 3 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension '3' priority 4 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension '3' priority 5 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension '3' priority 6 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension '3' priority 7 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension '4' priority 1 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension '4' priority 2 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension '4' priority 3 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension '4' priority 4 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension '4' priority 5 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension '4' priority 6 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension '4' priority 7 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension 'i' priority 1 to HUJI_IVR_MAIN_TAHZUKANIT  -- Added extension 'i' priority 2 to HUJI_IVR_MAIN_TAHZUKANIT [Sep 20 15:51:00] NOTICE[8200]: pbx_ael.c:1700 pbx_load_module: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'. [Sep 20 15:51:00] NOTICE[8200]: pbx_ael.c:1703 pbx_load_module: AEL load process: merged config file name '/etc/asterisk/extensions.ael'. [Sep 20 15:51:00] WARNING[8200]: pbx.c:6271 ast_context_verify_includes: Context 'huji-local' tries includes nonexistent context 'huji-local-hints' [Sep 20 15:51:00] NOTICE[8200]: pbx_ael.c:1706 pbx_load_module: AEL load process: verified config file name '/etc/asterisk/extensions.ael'.  pbx_ael.so => (Asterisk Extension Language Compiler)  == Registered custom function ICONV  func_iconv.so => (Charset conversions)  == Registered application 'Directory'  app_directory.so => (Extension Directory)  == Parsing '/etc/asterisk/codecs.conf':  == Found  -- codec_alaw: using generic PLC  == Registered translator 'alawtolin' from format alaw to slin, cost 85  == Registered translator 'lintoalaw' from format slin to alaw, cost 64  codec_alaw.so => (A-law Coder/Decoder)  == Registered file format wav49, extension(s) WAV|wav49  format_wav_gsm.so => (Microsoft WAV format (Proprietary GSM))  == Parsing '/etc/asterisk/oss.conf':  == Found  == Registered channel type 'Console' (OSS Console Channel Driver)  chan_oss.so => (OSS Console Channel Driver)  == Registered custom function CDR  func_cdr.so => (Call Detail Record (CDR) dialplan function)  == Registered custom function SHA1  func_sha1.so => (SHA-1 computation dialplan function)  pbx_realtime.so => (Realtime Switch)  == Registered application 'WaitForRing'  app_waitforring.so => (Waits until first ring after time)  == Registered application 'ForkCDR'  app_forkcdr.so => (Fork The CDR into 2 separate entities)  == Parsing '/etc/asterisk/codecs.conf':  == Found  -- codec_lpc10: using generic PLC  == Registered translator 'lpc10tolin' from format lpc10 to slin, cost 4397  == Registered translator 'lintolpc10' from format slin to lpc10, cost 5341  codec_lpc10.so => (LPC10 2.4kbps Coder/Decoder)  == Parsing '/etc/asterisk/codecs.conf':  == Found  -- codec_adpcm: using generic PLC  == Registered translator 'adpcmtolin' from format adpcm to slin, cost 365  == Registered translator 'lintoadpcm' from format slin to adpcm, cost 520  codec_adpcm.so => (Adaptive Differential PCM Coder/Decoder)  == Parsing '/etc/asterisk/codecs.conf':  == Found  -- codec_g722: using generic PLC  == Registered translator 'g722tolin' from format g722 to slin, cost 2994  == Registered translator 'lintog722' from format slin to g722, cost 3473  codec_g722.so => (ITU G.722-64kbps G722 Transcoder)  == Registered custom function GROUP_COUNT  == Registered custom function GROUP_MATCH_COUNT  == Registered custom function GROUP_LIST  == Registered custom function GROUP  func_groupcount.so => (Channel group dialplan functions)  == Parsing '/etc/asterisk/meetme.conf':  == Found  == Parsing '/etc/asterisk/sla.conf':  == Found  == Manager registered action MeetmeMute  == Manager registered action MeetmeUnmute  == Registered application 'MeetMeChannelAdmin'  == Registered application 'MeetMeAdmin'  == Registered application 'MeetMeCount'  == Registered application 'MeetMe'  == Registered application 'SLAStation'  == Registered application 'SLATrunk'  app_meetme.so => (MeetMe conference bridge)  == Registered application 'MixMonitor'  == Registered application 'StopMixMonitor'  app_mixmonitor.so => (Mixed Audio Monitoring Application)  -- Registered extension context 'app_dial_gosub_virtual_context'  -- Added extension 's' priority 1 to app_dial_gosub_virtual_context  == Registered application 'Dial'  == Registered application 'RetryDial'  app_dial.so => (Dialing Application)  == Registered application 'DumpChan'  app_dumpchan.so => (Dump Info About The Calling Channel)  == Registered application 'DBdel'  == Registered application 'DBdeltree'  app_db.so => (Database Access Functions)  == Registered file format vox, extension(s) vox  format_vox.so => (Dialogic VOX (ADPCM) File Format)  == Registered application 'MYSQL'  app_addon_sql_mysql.so => (Simple Mysql Interface)  == Parsing '/etc/asterisk/festival.conf':  == Found  == Registered application 'Festival'  app_festival.so => (Simple Festival Interface)  res_convert.so => (File format conversion CLI command)  == Registered application 'TestClient'  == Registered application 'TestServer'  app_test.so => (Interface Test Application)  == Parsing '/etc/asterisk/followme.conf':  == Found  == Registered application 'FollowMe'  app_followme.so => (Find-Me/Follow-Me Application)  == Registered application 'ReadFile'  app_readfile.so => (Stores output of file into a variable)  == Registered application 'ZapBarge'  app_zapbarge.so => (Barge in on Zap channel application)  == Registered application 'ChanIsAvail'  app_chanisavail.so => (Check channel availability)  == Registered custom function LOCK  == Registered custom function TRYLOCK  == Registered custom function UNLOCK  func_lock.so => (Dialplan mutexes)  == Parsing '/etc/asterisk/say.conf':  == Found  == Registered application 'Playback'  app_playback.so => (Sound File Playback Application)  == Registered application 'SoftHangup'  app_softhangup.so => (Hangs up the requested channel)  == Registered application 'BackgroundDetect'  app_talkdetect.so => (Playback with Talk Detection)  == Registered application 'SayUnixTime'  == Registered application 'DateTime'  app_sayunixtime.so => (Say time)  == Registered custom function DEVICE_STATE  == Registered custom function HINT  func_devstate.so => (Gets or sets a device state in the dialplan)  == Registered application 'Log'  == Registered application 'Verbose'  app_verbose.so => (Send verbose output)  == Registered application 'StackPop'  == Registered application 'Return'  == Registered application 'GosubIf'  == Registered application 'Gosub'  == Registered custom function LOCAL  app_stack.so => (Dialplan subroutines (Gosub, Return, etc))  == Registered application 'PrivacyManager'  app_privacy.so => (Require phone number to be entered, if no CallerID sent)  == Registered application 'ReadExten'  == Registered custom function VALID_EXTEN  app_readexten.so => (Read and evaluate extension validity)  == Registered application 'ICES'  app_ices.so => (Encode and Stream via icecast and ices)  == Parsing '/etc/asterisk/codecs.conf':  == Found  -- codec_g726: using generic PLC  == Registered translator 'g726tolin' from format g726 to slin, cost 3007  == Registered translator 'lintog726' from format slin to g726, cost 3168  == Registered translator 'g726aal2tolin' from format g726aal2 to slin, cost 2967  == Registered translator 'lintog726aal2' from format slin to g726aal2, cost 3171  == Registered translator 'g726aal2tog726' from format g726aal2 to g726, cost 98  == Registered translator 'g726tog726aal2' from format g726 to g726aal2, cost 97  codec_g726.so => (ITU G.726-32kbps G726 Transcoder)  == Manager registered action PlayDTMF  == Registered application 'SendDTMF'  app_senddtmf.so => (Send DTMF digits Application)  == Parsing '/etc/asterisk/dundi.conf':  == Found  == Parsing '/etc/asterisk/dundi.conf':  == Found  == Using TOS bits 0  == Using CoS mark 0  == Registered custom function DUNDILOOKUP  == Registered custom function DUNDIQUERY  == Registered custom function DUNDIRESULT  == DUNDi Ready and Listening on 0.0.0.0 port 4520  pbx_dundi.so => (Distributed Universal Number Discovery (DUNDi))  == Registered application 'ChanSpy'  == Registered application 'ExtenSpy'  app_chanspy.so => (Listen to the audio of an active channel)  == Registered application 'ParkAndAnnounce'  app_parkandannounce.so => (Call Parking and Announce Application)  == Parsing '/etc/asterisk/phone.conf':  == Found  == Registered channel type 'Phone' (Standard Linux Telephony API Driver)  chan_phone.so => (Linux Telephony API Support)  == Registered file format wav, extension(s) wav  format_wav.so => (Microsoft WAV format (8000Hz Signed Linear))  == Parsing '/etc/asterisk/mgcp.conf':  == Found  == MGCP Listening on 0.0.0.0:2727  == Using TOS bits 0  == Using CoS mark 0  == Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP))  chan_mgcp.so => (Media Gateway Control Protocol (MGCP))  == Registered custom function ENV  == Registered custom function STAT  func_env.so => (Environment/filesystem dialplan functions)  == Registered application 'Morsecode'  app_morsecode.so => (Morse code)  == Registered custom function VOLUME  func_volume.so => (Technology independent volume control)  == Registered application 'ExternalIVR'  app_externalivr.so => (External IVR Interface Application)  == Registered custom function BLACKLIST  func_blacklist.so => (Look up Caller*ID name/number from blacklist database)  == Registered custom function CALLERPRES  == Registered custom function CALLERID  func_callerid.so => (Caller ID related dialplan functions)  == Registered file format pcm, extension(s) pcm|ulaw|ul|mu  == Registered file format alaw, extension(s) alaw|al  == Registered file format au, extension(s) au  == Registered file format g722, extension(s) g722  format_pcm.so => (Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G.722 16Khz)  == Registered application 'Flash'  app_flash.so => (Flash channel application)  == Registered custom function BASE64_ENCODE  == Registered custom function BASE64_DECODE  func_base64.so => (base64 encode/decode dialplan functions)  == Registered application 'Transfer'  app_transfer.so => (Transfers a caller to another extension)  == Registered application 'Record'  app_record.so => (Trivial Record Application)  == Registered custom function MD5  func_md5.so => (MD5 digest dialplan functions)  == Registered application 'Read'  app_read.so => (Read Variable Application)  == Registered custom function EXTENSION_STATE  func_extstate.so => (Gets an extension's state in the dialplan)  == Registered file format g726-40, extension(s) g726-40  == Registered file format g726-32, extension(s) g726-32  == Registered file format g726-24, extension(s) g726-24  == Registered file format g726-16, extension(s) g726-16  format_g726.so => (Raw G.726 (16/24/32/40kbps) data)  == Registered custom function RAND  func_rand.so => (Random number dialplan function)  == Registered custom function FIELDQTY  == Registered custom function FILTER  == Registered custom function REGEX  == Registered custom function ARRAY  == Registered custom function QUOTE  == Registered custom function LEN  == Registered custom function STRFTIME  == Registered custom function STRPTIME  == Registered custom function EVAL  == Registered custom function KEYPADHASH  == Registered custom function SPRINTF  == Registered custom function HASHKEYS  == Registered custom function HASH  == Registered application 'ClearHash'  func_strings.so => (String handling dialplan functions)  == Parsing '/etc/asterisk/extensions.conf':  == Found  == Setting global variable 'CONSOLE' to 'Console/dsp'  -- Registered extension context 'huji-local-hints'  -- Added extension '80600' priority -1 to huji-local-hints  -- Added extension '80601' priority -1 to huji-local-hints  -- Added extension '80602' priority -1 to huji-local-hints  -- Added extension '80603' priority -1 to huji-local-hints  -- Added extension '80604' priority -1 to huji-local-hints  -- Added extension '80605' priority -1 to huji-local-hints  -- Added extension '80606' priority -1 to huji-local-hints  -- Added extension '80607' priority -1 to huji-local-hints  -- Added extension '80608' priority -1 to huji-local-hints  -- Added extension '80609' priority -1 to huji-local-hints  -- Added extension '80610' priority -1 to huji-local-hints  -- Added extension '80611' priority -1 to huji-local-hints  -- Added extension '80612' priority -1 to huji-local-hints  -- Added extension '80613' priority -1 to huji-local-hints  -- Added extension '80614' priority -1 to huji-local-hints  -- Added extension '80615' priority -1 to huji-local-hints  -- Added extension '80616' priority -1 to huji-local-hints  -- Added extension '80617' priority -1 to huji-local-hints  -- Added extension '80618' priority -1 to huji-local-hints  -- Added extension '80619' priority -1 to huji-local-hints  -- Added extension '80620' priority -1 to huji-local-hints  -- Added extension '80622' priority -1 to huji-local-hints  -- Added extension '80680' priority -1 to huji-local-hints  -- Added extension '82739' priority -1 to huji-local-hints  -- Added extension '84138' priority -1 to huji-local-hints  -- Added extension '84413' priority -1 to huji-local-hints  -- Added extension '84280' priority -1 to huji-local-hints  -- Added extension '86244' priority -1 to huji-local-hints  -- Added extension '84989' priority -1 to huji-local-hints  -- Added extension '89444' priority -1 to huji-local-hints  == Parsing '/etc/asterisk/users.conf':  == Found  -- Registered extension context 'default'  pbx_config.so => (Text Extension Configuration)  == Registered channel type 'Agent' (Call Agent Proxy Channel)  == Parsing '/etc/asterisk/agents.conf':  == Found  == Parsing '/etc/asterisk/users.conf':  == Found  == Registered application 'AgentLogin'  == Registered application 'AgentMonitorOutgoing'  == Manager registered action Agents  == Manager registered action AgentLogoff  == Registered custom function AGENT  chan_agent.so => (Agent Proxy Channel)  == Registered custom function MATH  func_math.so => (Mathematical dialplan function)  == Registered custom function VERSION  func_version.so => (Get Asterisk Version/Build Info)  == Parsing '/etc/asterisk/queues.conf':  == Found  -- Registered extension context 'app_queue_gosub_virtual_context'  -- Added extension 's' priority 1 to app_queue_gosub_virtual_context  == Registered application 'Queue'  == Registered application 'AddQueueMember'  == Registered application 'RemoveQueueMember'  == Registered application 'PauseQueueMember'  == Registered application 'UnpauseQueueMember'  == Registered application 'QueueLog'  == Manager registered action Queues  == Manager registered action QueueStatus  == Manager registered action QueueSummary  == Manager registered action QueueAdd  == Manager registered action QueueRemove  == Manager registered action QueuePause  == Manager registered action QueueLog  == Registered custom function QUEUE_VARIABLES  == Registered custom function QUEUE_MEMBER_COUNT  == Registered custom function QUEUE_MEMBER_LIST  == Registered custom function QUEUE_WAITING_COUNT  app_queue.so => (True Call Queueing)  == Registered application 'MacroExit'  == Registered application 'MacroIf'  == Registered application 'MacroExclusive'  == Registered application 'Macro'  app_macro.so => (Extension Macros)  == Registered custom function VMCOUNT  func_vmcount.so => (Indicator for whether a voice mailbox has messages in a given folder.)  == Registered custom function SHELL  func_shell.so => (Returns the output of a shell command)  == Registered file format gsm, extension(s) gsm  format_gsm.so => (Raw GSM data)  res_realtime.so => (Realtime Data Lookup/Rewrite)  == Registered custom function REALTIME  func_realtime.so => (Read/Write values from a RealTime repository)  == Registered application 'SetCallerPres'  app_setcallerid.so => (Set CallerID Presentation Application)  == Registered application 'Echo'  app_echo.so => (Simple Echo Application)  res_clioriginate.so => (Call origination from the CLI) Asterisk Ready. *CLI> [Sep 20 15:51:07] NOTICE[8200]: chan_sip.c:15827 handle_request_subscribe: Failed to authenticate user ;tag=efuj7rvebm for SUBSCRIBE [Sep 20 15:51:07] NOTICE[8200]: chan_sip.c:15827 handle_request_subscribe: Failed to authenticate user ;tag=efuj7rvebm for SUBSCRIBE  -- Unregistered SIP '85684' sip se sip set debug on SIP Debugging enabled *CLI> Retransmitting #5 (NAT) to 132.64.4.137:2132: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 132.64.4.137:2132;branch=z9hG4bK-v4ft73mzgqba;received=132.64.4.137;rport=2132 From: ;tag=efuj7rvebm To: ;tag=as2d84524b Call-ID: 3c26700a1388-wjx5tt2ysc3u@snom320-0004132480B4 CSeq: 4 SUBSCRIBE User-Agent: Asterisk PBX SVN-trunk-r83213M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Retransmitting #5 (NAT) to 132.64.4.137:2132: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 132.64.4.137:2132;branch=z9hG4bK-l7klxvv8rfoa;received=132.64.4.137;rport=2132 From: ;tag=efuj7rvebm To: ;tag=as2d84524b Call-ID: 3c26700a1388-wjx5tt2ysc3u@snom320-0004132480B4 CSeq: 5 SUBSCRIBE User-Agent: Asterisk PBX SVN-trunk-r83213M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Retransmitting #6 (NAT) to 132.64.4.137:2132: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 132.64.4.137:2132;branch=z9hG4bK-v4ft73mzgqba;received=132.64.4.137;rport=2132 From: ;tag=efuj7rvebm To: ;tag=as2d84524b Call-ID: 3c26700a1388-wjx5tt2ysc3u@snom320-0004132480B4 CSeq: 4 SUBSCRIBE User-Agent: Asterisk PBX SVN-trunk-r83213M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Retransmitting #6 (NAT) to 132.64.4.137:2132: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 132.64.4.137:2132;branch=z9hG4bK-l7klxvv8rfoa;received=132.64.4.137;rport=2132 From: ;tag=efuj7rvebm To: ;tag=as2d84524b Call-ID: 3c26700a1388-wjx5tt2ysc3u@snom320-0004132480B4 CSeq: 5 SUBSCRIBE User-Agent: Asterisk PBX SVN-trunk-r83213M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- <--- SIP read from 132.64.4.137:2135 ---> REGISTER sip:pbx-dev.cc.huji.ac.il SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2135;branch=z9hG4bK-ltwmw7nljquf;rport From: ;tag=2l0cf1x9w0 To: Call-ID: 3c26700a493e-sbiow7lwpu03@snom320-0004132480B4 CSeq: 1 REGISTER Max-Forwards: 70 Contact: ;q=1.0;flow-id=1;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom320";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom320/6.2.2 Supported: gruu Allow-Events: dialog X-Real-IP: 132.64.4.137 WWW-Contact: Expires: 3600 Content-Length: 0 <-------------> --- (15 headers 0 lines) --- Sending to 132.64.4.137 : 2135 (NAT) <--- Transmitting (no NAT) to 132.64.4.137:2135 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 132.64.4.137:2135;branch=z9hG4bK-ltwmw7nljquf;received=132.64.4.137;rport=2135 From: ;tag=2l0cf1x9w0 To: ;tag=as6df77fb8 Call-ID: 3c26700a493e-sbiow7lwpu03@snom320-0004132480B4 CSeq: 1 REGISTER User-Agent: Asterisk PBX SVN-trunk-r83213M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="cc.huji.ac.il", nonce="1afe67bd" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3c26700a493e-sbiow7lwpu03@snom320-0004132480B4' in 32000 ms (Method: REGISTER) <--- SIP read from 132.64.4.137:2135 ---> SUBSCRIBE sip:80620@pbx-dev.cc.huji.ac.il;user=phone SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2135;branch=z9hG4bK-3vhk6hianhcb;rport From: ;tag=cypxxhdq5i To: Call-ID: 3c26700a445c-wjx5tt2ysc3u@snom320-0004132480B4 CSeq: 1 SUBSCRIBE Max-Forwards: 70 Contact: ;flow-id=1 Event: dialog Accept: application/dialog-info+xml Expires: 3600 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Creating new subscription Sending to 132.64.4.137 : 2135 (NAT) No matching peer for '89444' from '132.64.4.137:2135' [Sep 20 15:51:23] NOTICE[8200]: chan_sip.c:15827 handle_request_subscribe: Failed to authenticate user ;tag=cypxxhdq5i for SUBSCRIBE <--- Reliably Transmitting (NAT) to 132.64.4.137:2135 ---> SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 132.64.4.137:2135;branch=z9hG4bK-3vhk6hianhcb;received=132.64.4.137;rport=2135 From: ;tag=cypxxhdq5i To: ;tag=as79f54839 Call-ID: 3c26700a445c-wjx5tt2ysc3u@snom320-0004132480B4 CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX SVN-trunk-r83213M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> <--- SIP read from 132.64.4.137:2135 ---> REGISTER sip:pbx-dev.cc.huji.ac.il SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2135;branch=z9hG4bK-nyeu4x72xuj2;rport From: ;tag=2l0cf1x9w0 To: Call-ID: 3c26700a493e-sbiow7lwpu03@snom320-0004132480B4 CSeq: 2 REGISTER Max-Forwards: 70 Contact: ;q=1.0;flow-id=1;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom320";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom320/6.2.2 Supported: gruu Allow-Events: dialog X-Real-IP: 132.64.4.137 WWW-Contact: Authorization: Digest username="85684",realm="cc.huji.ac.il",nonce="1afe67bd",uri="sip:pbx-dev.cc.huji.ac.il",response="e16f1df82da96e1d87f4885eee814f72",algorithm=md5 Expires: 3600 Content-Length: 0 <-------------> --- (16 headers 0 lines) --- Sending to 132.64.4.137 : 2135 (NAT)  -- Registered SIP '85684' at 132.64.4.137 port 2135 expires 3600  > Saved useragent "snom320/6.2.2" for peer 85684 <--- Transmitting (no NAT) to 132.64.4.137:2135 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.4.137:2135;branch=z9hG4bK-nyeu4x72xuj2;received=132.64.4.137;rport=2135 From: ;tag=2l0cf1x9w0 To: ;tag=as6df77fb8 Call-ID: 3c26700a493e-sbiow7lwpu03@snom320-0004132480B4 CSeq: 2 REGISTER User-Agent: Asterisk PBX SVN-trunk-r83213M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 3600 Contact: ;expires=3600 Date: Thu, 20 Sep 2007 13:51:24 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3c26700a493e-sbiow7lwpu03@snom320-0004132480B4' in 32000 ms (Method: REGISTER) <--- SIP read from 132.64.4.137:2135 ---> SUBSCRIBE sip:80622@pbx-dev.cc.huji.ac.il SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2135;branch=z9hG4bK-yljtt7535ou0;rport From: ;tag=nju6ht3sh5 To: Call-ID: 3c26700a6ddd-0gsczhjqwmf1@snom320-0004132480B4 CSeq: 1 SUBSCRIBE Max-Forwards: 70 Contact: ;flow-id=1 Event: dialog Accept: application/dialog-info+xml Expires: 3600 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Creating new subscription Sending to 132.64.4.137 : 2135 (NAT) Found peer '85684' for '85684' from 132.64.4.137:2135 <--- Transmitting (no NAT) to 132.64.4.137:2135 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 132.64.4.137:2135;branch=z9hG4bK-yljtt7535ou0;received=132.64.4.137;rport=2135 From: ;tag=nju6ht3sh5 To: ;tag=as4586c478 Call-ID: 3c26700a6ddd-0gsczhjqwmf1@snom320-0004132480B4 CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX SVN-trunk-r83213M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="cc.huji.ac.il", nonce="0f009449" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3c26700a6ddd-0gsczhjqwmf1@snom320-0004132480B4' in 32000 ms (Method: SUBSCRIBE) <--- SIP read from 132.64.4.137:2135 ---> SUBSCRIBE sip:84138@pbx-dev.cc.huji.ac.il SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2135;branch=z9hG4bK-k3xtk1i16079;rport From: ;tag=zl3fa48osv To: Call-ID: 3c26700a6ddd-vj6y9ke9ouf1@snom320-0004132480B4 CSeq: 1 SUBSCRIBE Max-Forwards: 70 Contact: ;flow-id=1 Event: dialog Accept: application/dialog-info+xml Expires: 3600 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Creating new subscription Sending to 132.64.4.137 : 2135 (NAT) Found peer '85684' for '85684' from 132.64.4.137:2135 <--- Transmitting (no NAT) to 132.64.4.137:2135 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 132.64.4.137:2135;branch=z9hG4bK-k3xtk1i16079;received=132.64.4.137;rport=2135 From: ;tag=zl3fa48osv To: ;tag=as230b6ce8 Call-ID: 3c26700a6ddd-vj6y9ke9ouf1@snom320-0004132480B4 CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX SVN-trunk-r83213M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="cc.huji.ac.il", nonce="4da278bc" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3c26700a6ddd-vj6y9ke9ouf1@snom320-0004132480B4' in 32000 ms (Method: SUBSCRIBE) <--- SIP read from 132.64.4.137:2135 ---> SUBSCRIBE sip:80620@pbx-dev.cc.huji.ac.il SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2135;branch=z9hG4bK-ct1yttby4v11;rport From: ;tag=w0gasmyayd To: Call-ID: 3c26700a6ddd-7f3p477qk4w1@snom320-0004132480B4 CSeq: 1 SUBSCRIBE Max-Forwards: 70 Contact: ;flow-id=1 Event: dialog Accept: application/dialog-info+xml Expires: 3600 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Creating new subscription Sending to 132.64.4.137 : 2135 (NAT) Found peer '85684' for '85684' from 132.64.4.137:2135 <--- Transmitting (no NAT) to 132.64.4.137:2135 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 132.64.4.137:2135;branch=z9hG4bK-ct1yttby4v11;received=132.64.4.137;rport=2135 From: ;tag=w0gasmyayd To: ;tag=as2b663d6f Call-ID: 3c26700a6ddd-7f3p477qk4w1@snom320-0004132480B4 CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX SVN-trunk-r83213M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="cc.huji.ac.il", nonce="46a9df83" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3c26700a6ddd-7f3p477qk4w1@snom320-0004132480B4' in 32000 ms (Method: SUBSCRIBE) <--- SIP read from 132.64.4.137:2135 ---> SUBSCRIBE sip:80622@pbx-dev.cc.huji.ac.il SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2135;branch=z9hG4bK-lmqmtadmnynp;rport From: ;tag=nju6ht3sh5 To: Call-ID: 3c26700a6ddd-0gsczhjqwmf1@snom320-0004132480B4 CSeq: 2 SUBSCRIBE Max-Forwards: 70 Contact: ;flow-id=1 Event: dialog Accept: application/dialog-info+xml Authorization: Digest username="85684",realm="cc.huji.ac.il",nonce="0f009449",uri="sip:80622@pbx-dev.cc.huji.ac.il",response="0ac9bef5879d57b5df46d20c46f5a957",algorithm=md5 Expires: 3600 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Creating new subscription Sending to 132.64.4.137 : 2135 (NAT) Found peer '85684' for '85684' from 132.64.4.137:2135 Looking for 80622 in huji-remote-gr (domain pbx-dev.cc.huji.ac.il) Scheduling destruction of SIP dialog '3c26700a6ddd-0gsczhjqwmf1@snom320-0004132480B4' in 3610000 ms (Method: SUBSCRIBE) <--- Transmitting (no NAT) to 132.64.4.137:2135 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.4.137:2135;branch=z9hG4bK-lmqmtadmnynp;received=132.64.4.137;rport=2135 From: ;tag=nju6ht3sh5 To: ;tag=as4586c478 Call-ID: 3c26700a6ddd-0gsczhjqwmf1@snom320-0004132480B4 CSeq: 2 SUBSCRIBE User-Agent: Asterisk PBX SVN-trunk-r83213M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 3600 Contact: ;expires=3600 Content-Length: 0 <------------> failed to extend from 0 to 44 failed to extend from 0 to 171 failed to extend from 0 to 42 failed to extend from 0 to 49 failed to extend from 0 to 48 Reliably Transmitting (no NAT) to 132.64.4.137:2135: NOTIFY sip:85684@132.64.4.137:2135;line=mpqvu5xe SIP/2.0 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK7fa71cff;rport Max-Forwards: 70 From: ;tag=as4586c478 To: ;tag=nju6ht3sh5 Contact: Call-ID: 3c26700a6ddd-0gsczhjqwmf1@snom320-0004132480B4 CSeq: 102 NOTIFY User-Agent: Asterisk PBX SVN-trunk-r83213M Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 0 --- <--- SIP read from 132.64.4.137:2135 ---> SUBSCRIBE sip:84138@pbx-dev.cc.huji.ac.il SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2135;branch=z9hG4bK-tgo3pkhaja3u;rport From: ;tag=zl3fa48osv To: Call-ID: 3c26700a6ddd-vj6y9ke9ouf1@snom320-0004132480B4 CSeq: 2 SUBSCRIBE Max-Forwards: 70 Contact: ;flow-id=1 Event: dialog Accept: application/dialog-info+xml Authorization: Digest username="85684",realm="cc.huji.ac.il",nonce="4da278bc",uri="sip:84138@pbx-dev.cc.huji.ac.il",response="00185bdc6239597916de01715f4b4ce1",algorithm=md5 Expires: 3600 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Creating new subscription Sending to 132.64.4.137 : 2135 (NAT) Found peer '85684' for '85684' from 132.64.4.137:2135 Looking for 84138 in huji-remote-gr (domain pbx-dev.cc.huji.ac.il) Scheduling destruction of SIP dialog '3c26700a6ddd-vj6y9ke9ouf1@snom320-0004132480B4' in 3610000 ms (Method: SUBSCRIBE) <--- Transmitting (no NAT) to 132.64.4.137:2135 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.4.137:2135;branch=z9hG4bK-tgo3pkhaja3u;received=132.64.4.137;rport=2135 From: ;tag=zl3fa48osv To: ;tag=as230b6ce8 Call-ID: 3c26700a6ddd-vj6y9ke9ouf1@snom320-0004132480B4 CSeq: 2 SUBSCRIBE User-Agent: Asterisk PBX SVN-trunk-r83213M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 3600 Contact: ;expires=3600 Content-Length: 0 <------------> failed to extend from 0 to 44 failed to extend from 0 to 171 failed to extend from 0 to 42 failed to extend from 0 to 49 failed to extend from 0 to 48 Reliably Transmitting (no NAT) to 132.64.4.137:2135: NOTIFY sip:85684@132.64.4.137:2135;line=mpqvu5xe SIP/2.0 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK68c05d29;rport Max-Forwards: 70 From: ;tag=as230b6ce8 To: ;tag=zl3fa48osv Contact: Call-ID: 3c26700a6ddd-vj6y9ke9ouf1@snom320-0004132480B4 CSeq: 102 NOTIFY User-Agent: Asterisk PBX SVN-trunk-r83213M Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 0 --- <--- SIP read from 132.64.4.137:2135 ---> SUBSCRIBE sip:80620@pbx-dev.cc.huji.ac.il SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2135;branch=z9hG4bK-r99thefslsmy;rport From: ;tag=w0gasmyayd To: Call-ID: 3c26700a6ddd-7f3p477qk4w1@snom320-0004132480B4 CSeq: 2 SUBSCRIBE Max-Forwards: 70 Contact: ;flow-id=1 Event: dialog Accept: application/dialog-info+xml Authorization: Digest username="85684",realm="cc.huji.ac.il",nonce="46a9df83",uri="sip:80620@pbx-dev.cc.huji.ac.il",response="dab8c2dc94d9c145c886856d4e9861cf",algorithm=md5 Expires: 3600 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Creating new subscription Sending to 132.64.4.137 : 2135 (NAT) Found peer '85684' for '85684' from 132.64.4.137:2135 Looking for 80620 in huji-remote-gr (domain pbx-dev.cc.huji.ac.il) Scheduling destruction of SIP dialog '3c26700a6ddd-7f3p477qk4w1@snom320-0004132480B4' in 3610000 ms (Method: SUBSCRIBE) <--- Transmitting (no NAT) to 132.64.4.137:2135 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.4.137:2135;branch=z9hG4bK-r99thefslsmy;received=132.64.4.137;rport=2135 From: ;tag=w0gasmyayd To: ;tag=as2b663d6f Call-ID: 3c26700a6ddd-7f3p477qk4w1@snom320-0004132480B4 CSeq: 2 SUBSCRIBE User-Agent: Asterisk PBX SVN-trunk-r83213M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 3600 Contact: ;expires=3600 Content-Length: 0 <------------> failed to extend from 0 to 44 failed to extend from 0 to 171 failed to extend from 0 to 42 failed to extend from 0 to 49 failed to extend from 0 to 48 Reliably Transmitting (no NAT) to 132.64.4.137:2135: NOTIFY sip:85684@132.64.4.137:2135;line=mpqvu5xe SIP/2.0 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK1a67746c;rport Max-Forwards: 70 From: ;tag=as2b663d6f To: ;tag=w0gasmyayd Contact: Call-ID: 3c26700a6ddd-7f3p477qk4w1@snom320-0004132480B4 CSeq: 102 NOTIFY User-Agent: Asterisk PBX SVN-trunk-r83213M Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 0 --- <--- SIP read from 132.64.4.137:2135 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK7fa71cff;rport=5060 From: ;tag=as4586c478 To: ;tag=nju6ht3sh5 Call-ID: 3c26700a6ddd-0gsczhjqwmf1@snom320-0004132480B4 CSeq: 102 NOTIFY Content-Length: 0 <-------------> --- (7 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 132.64.4.137:2135 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK68c05d29;rport=5060 From: ;tag=as230b6ce8 To: ;tag=zl3fa48osv Call-ID: 3c26700a6ddd-vj6y9ke9ouf1@snom320-0004132480B4 CSeq: 102 NOTIFY Content-Length: 0 <-------------> --- (7 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 132.64.4.137:2135 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK1a67746c;rport=5060 From: ;tag=as2b663d6f To: ;tag=w0gasmyayd Call-ID: 3c26700a6ddd-7f3p477qk4w1@snom320-0004132480B4 CSeq: 102 NOTIFY Content-Length: 0 <-------------> --- (7 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived Retransmitting #1 (NAT) to 132.64.4.137:2135: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 132.64.4.137:2135;branch=z9hG4bK-3vhk6hianhcb;received=132.64.4.137;rport=2135 From: ;tag=cypxxhdq5i To: ;tag=as79f54839 Call-ID: 3c26700a445c-wjx5tt2ysc3u@snom320-0004132480B4 CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX SVN-trunk-r83213M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- <--- SIP read from 132.64.4.120:5060 ---> REGISTER sip:pbx-dev.cc.huji.ac.il SIP/2.0 Via: SIP/2.0/UDP 132.64.4.120:5060;branch=z9hG4bKa6fea3b9ff2efe7c From: Line1 ATAtest ;tag=3574685101 To: Line1 ATAtest Call-ID: 2843921503@132.64.4.120 CSeq: 70 REGISTER Contact: Line1 ATAtest ;expires=900 User-Agent: Cisco ATA 186 v3.2.0 atasip (041111A) Authorization: Digest username="80681",realm="cc.huji.ac.il",nonce="288327f2",uri="sip:pbx-dev.cc.huji.ac.il",response="d2fe77be7c17bdea0eb84f2f466cd3a2" Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 132.64.4.120 : 5060 (no NAT) <--- Transmitting (no NAT) to 132.64.4.120:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 132.64.4.120:5060;branch=z9hG4bKa6fea3b9ff2efe7c;received=132.64.4.120 From: Line1 ATAtest ;tag=3574685101 To: Line1 ATAtest ;tag=as28cc7bf0 Call-ID: 2843921503@132.64.4.120 CSeq: 70 REGISTER User-Agent: Asterisk PBX SVN-trunk-r83213M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="cc.huji.ac.il", nonce="46ccff21" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '2843921503@132.64.4.120' in 32000 ms (Method: REGISTER) <--- SIP read from 132.64.4.120:5060 ---> REGISTER sip:pbx-dev.cc.huji.ac.il SIP/2.0 Via: SIP/2.0/UDP 132.64.4.120:5060;branch=z9hG4bKf6c68a07e7ada3ae From: Line1 ATAtest ;tag=3574685101 To: Line1 ATAtest Call-ID: 2843921503@132.64.4.120 CSeq: 71 REGISTER Contact: Line1 ATAtest ;expires=900 User-Agent: Cisco ATA 186 v3.2.0 atasip (041111A) Authorization: Digest username="80681",realm="cc.huji.ac.il",nonce="46ccff21",uri="sip:pbx-dev.cc.huji.ac.il",response="0cc3ab296bed5f2af5af4234995280fa" Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 132.64.4.120 : 5060 (no NAT) <--- Transmitting (no NAT) to 132.64.4.120:5060 ---> SIP/2.0 403 Forbidden (Bad auth) Via: SIP/2.0/UDP 132.64.4.120:5060;branch=z9hG4bKf6c68a07e7ada3ae;received=132.64.4.120 From: Line1 ATAtest ;tag=3574685101 To: Line1 ATAtest ;tag=as28cc7bf0 Call-ID: 2843921503@132.64.4.120 CSeq: 71 REGISTER User-Agent: Asterisk PBX SVN-trunk-r83213M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> [Sep 20 15:51:25] NOTICE[8200]: chan_sip.c:16082 handle_request_register: Registration from 'Line1 ATAtest ' failed for '132.64.4.120' - Wrong password Scheduling destruction of SIP dialog '2843921503@132.64.4.120' in 32000 ms (Method: REGISTER) Retransmitting #2 (NAT) to 132.64.4.137:2135: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 132.64.4.137:2135;branch=z9hG4bK-3vhk6hianhcb;received=132.64.4.137;rport=2135 From: ;tag=cypxxhdq5i To: ;tag=as79f54839 Call-ID: 3c26700a445c-wjx5tt2ysc3u@snom320-0004132480B4 CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX SVN-trunk-r83213M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Sep 20 15:51:27] WARNING[8200]: chan_sip.c:2299 retrans_pkt: Maximum retries exceeded on transmission 3c26700a1388-wjx5tt2ysc3u@snom320-0004132480B4 for seqno 4 (Critical Response) [Sep 20 15:51:27] WARNING[8200]: chan_sip.c:2299 retrans_pkt: Maximum retries exceeded on transmission 3c26700a1388-wjx5tt2ysc3u@snom320-0004132480B4 for seqno 5 (Critical Response) Really destroying SIP dialog '3c26700a1388-wjx5tt2ysc3u@snom320-0004132480B4' Method: SUBSCRIBE Retransmitting #3 (NAT) to 132.64.4.137:2135: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 132.64.4.137:2135;branch=z9hG4bK-3vhk6hianhcb;received=132.64.4.137;rport=2135 From: ;tag=cypxxhdq5i To: ;tag=as79f54839 Call-ID: 3c26700a445c-wjx5tt2ysc3u@snom320-0004132480B4 CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX SVN-trunk-r83213M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- <--- SIP read from 132.64.4.120:5060 ---> REGISTER sip:pbx-dev.cc.huji.ac.il SIP/2.0 Via: SIP/2.0/UDP 132.64.4.120:5060;branch=z9hG4bK4be2bf6d22ba39d2 From: line2 ATAtest ;tag=492216117 To: line2 ATAtest Call-ID: 2222414595@132.64.4.120 CSeq: 10 REGISTER Contact: line2 ATAtest ;expires=900 User-Agent: Cisco ATA 186 v3.2.0 atasip (041111A) Authorization: Digest username="80611",realm="cc.huji.ac.il",nonce="624ef608",uri="sip:pbx-dev.cc.huji.ac.il",response="f7618c002be9a74cc98a5753e507ae8c" Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 132.64.4.120 : 5060 (no NAT) <--- Transmitting (no NAT) to 132.64.4.120:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 132.64.4.120:5060;branch=z9hG4bK4be2bf6d22ba39d2;received=132.64.4.120 From: line2 ATAtest ;tag=492216117 To: line2 ATAtest ;tag=as64b4fbba Call-ID: 2222414595@132.64.4.120 CSeq: 10 REGISTER User-Agent: Asterisk PBX SVN-trunk-r83213M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="cc.huji.ac.il", nonce="5866aae4" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '2222414595@132.64.4.120' in 32000 ms (Method: REGISTER) <--- SIP read from 132.64.4.120:5060 ---> REGISTER sip:pbx-dev.cc.huji.ac.il SIP/2.0 Via: SIP/2.0/UDP 132.64.4.120:5060;branch=z9hG4bK8821de53d479e7cc From: line2 ATAtest ;tag=492216117 To: line2 ATAtest Call-ID: 2222414595@132.64.4.120 CSeq: 11 REGISTER Contact: line2 ATAtest ;expires=900 User-Agent: Cisco ATA 186 v3.2.0 atasip (041111A) Authorization: Digest username="80611",realm="cc.huji.ac.il",nonce="5866aae4",uri="sip:pbx-dev.cc.huji.ac.il",response="5c6409e6ddd2920cdf14ace872ad8256" Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 132.64.4.120 : 5060 (no NAT)  > Saved useragent "Cisco ATA 186 v3.2.0 atasip (041111A)" for peer 80611 <--- Transmitting (no NAT) to 132.64.4.120:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.4.120:5060;branch=z9hG4bK8821de53d479e7cc;received=132.64.4.120 From: line2 ATAtest ;tag=492216117 To: line2 ATAtest ;tag=as64b4fbba Call-ID: 2222414595@132.64.4.120 CSeq: 11 REGISTER User-Agent: Asterisk PBX SVN-trunk-r83213M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 900 Contact: ;expires=900 Date: Thu, 20 Sep 2007 13:51:30 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '2222414595@132.64.4.120' in 32000 ms (Method: REGISTER) Retransmitting #4 (NAT) to 132.64.4.137:2135: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 132.64.4.137:2135;branch=z9hG4bK-3vhk6hianhcb;received=132.64.4.137;rport=2135 From: ;tag=cypxxhdq5i To: ;tag=as79f54839 Call-ID: 3c26700a445c-wjx5tt2ysc3u@snom320-0004132480B4 CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX SVN-trunk-r83213M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Retransmitting #5 (NAT) to 132.64.4.137:2135: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 132.64.4.137:2135;branch=z9hG4bK-3vhk6hianhcb;received=132.64.4.137;rport=2135 From: ;tag=cypxxhdq5i To: ;tag=as79f54839 Call-ID: 3c26700a445c-wjx5tt2ysc3u@snom320-0004132480B4 CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX SVN-trunk-r83213M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- <--- SIP read from 132.64.4.120:5060 ---> INVITE sip:80620@pbx-dev.cc.huji.ac.il;user=phone SIP/2.0 Via: SIP/2.0/UDP 132.64.4.120:5060;branch=z9hG4bK3c8edbfa1ed5db91 From: line2 ATAtest ;tag=951678558 To: Call-ID: 277848150@132.64.4.120 CSeq: 1 INVITE Contact: line2 ATAtest User-Agent: Cisco ATA 186 v3.2.0 atasip (041111A) Expires: 300 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: 100rel,replaces Content-Length: 226 Content-Type: application/sdp v=0 o=80611 3304609 3304609 IN IP4 132.64.4.120 s=ATA186 Call c=IN IP4 132.64.4.120 t=0 0 m=audio 16384 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (13 headers 10 lines) ---  == Using TOS bits 0  == Using CoS mark 5  == Using TOS bits 0  == Using CoS mark 6  == Using TOS bits 0  == Using CoS mark 5 Sending to 132.64.4.120 : 5060 (no NAT) Using INVITE request as basis request - 277848150@132.64.4.120 Found user '80611' for '80611' <--- Reliably Transmitting (no NAT) to 132.64.4.120:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 132.64.4.120:5060;branch=z9hG4bK3c8edbfa1ed5db91;received=132.64.4.120 From: line2 ATAtest ;tag=951678558 To: ;tag=as536e5f10 Call-ID: 277848150@132.64.4.120 CSeq: 1 INVITE User-Agent: Asterisk PBX SVN-trunk-r83213M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="cc.huji.ac.il", nonce="528091f4" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '277848150@132.64.4.120' in 32000 ms (Method: INVITE) <--- SIP read from 132.64.4.120:5060 ---> ACK sip:80620@pbx-dev.cc.huji.ac.il;user=phone SIP/2.0 Via: SIP/2.0/UDP 132.64.4.120:5060;branch=z9hG4bK3c8edbfa1ed5db91 From: line2 ATAtest ;tag=951678558 To: ;tag=as536e5f10 Call-ID: 277848150@132.64.4.120 CSeq: 1 ACK User-Agent: Cisco ATA 186 v3.2.0 atasip (041111A) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 132.64.4.120:5060 ---> INVITE sip:80620@pbx-dev.cc.huji.ac.il;user=phone SIP/2.0 Via: SIP/2.0/UDP 132.64.4.120:5060;branch=z9hG4bKa61b0b03ee7aad0d From: line2 ATAtest ;tag=951678558 To: Call-ID: 277848150@132.64.4.120 CSeq: 2 INVITE Contact: line2 ATAtest User-Agent: Cisco ATA 186 v3.2.0 atasip (041111A) Authorization: Digest username="80611",realm="cc.huji.ac.il",nonce="528091f4",uri="sip:80620@pbx-dev.cc.huji.ac.il",response="4db55781d7c1d04a71db8510e1d5d010" Expires: 300 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: 100rel,replaces Content-Length: 226 Content-Type: application/sdp v=0 o=80611 3304613 3304613 IN IP4 132.64.4.120 s=ATA186 Call c=IN IP4 132.64.4.120 t=0 0 m=audio 16384 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (14 headers 10 lines) --- Sending to 132.64.4.120 : 5060 (no NAT) Using INVITE request as basis request - 277848150@132.64.4.120 Found user '80611' for '80611' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 132.64.4.120:16384 Found description format PCMA for ID 8 Found description format PCMU for ID 0 Found description format telephone-event for ID 101 Capabilities: us - 0xc040e (gsm|ulaw|alaw|ilbc|h261|h263), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 132.64.4.120:16384 Looking for 80620 in huji-remote-gr (domain pbx-dev.cc.huji.ac.il) list_route: hop: <--- Transmitting (no NAT) to 132.64.4.120:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 132.64.4.120:5060;branch=z9hG4bKa61b0b03ee7aad0d;received=132.64.4.120 From: line2 ATAtest ;tag=951678558 To: Call-ID: 277848150@132.64.4.120 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r83213M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------>  -- Executing [80620@huji-remote-gr:1] NoOp("SIP/80611-b7d56a88", "") in new stack  -- Executing [80620@huji-remote-gr:2] Set("SIP/80611-b7d56a88", "_To=80620") in new stack  -- Executing [80620@huji-remote-gr:3] Set("SIP/80611-b7d56a88", "_From=80611") in new stack  -- Executing [80620@huji-remote-gr:4] Set("SIP/80611-b7d56a88", "DB(80620/LastCaller)=80611") in new stack  -- Executing [80620@huji-remote-gr:5] Set("SIP/80611-b7d56a88", "DB(80611/LastCalled)=80620") in new stack  -- Executing [80620@huji-remote-gr:6] MYSQL("SIP/80611-b7d56a88", "Connect connid localhost asterisk NotMe43e080 asterisk") in new stack  -- Executing [80620@huji-remote-gr:7] MYSQL("SIP/80611-b7d56a88", "Query resID 1 SELECT additional_numbers from shared_lines where orig_number='80620'") in new stack  -- Executing [80620@huji-remote-gr:8] MYSQL("SIP/80611-b7d56a88", "Fetch FetchId 2 aEXTEN") in new stack  -- Executing [80620@huji-remote-gr:9] NoOp("SIP/80611-b7d56a88", "") in new stack  -- Executing [80620@huji-remote-gr:10] MYSQL("SIP/80611-b7d56a88", "Clear 2") in new stack  -- Executing [80620@huji-remote-gr:11] MYSQL("SIP/80611-b7d56a88", "Query resID 1 SELECT callerid from sip_users where name='80620'") in new stack  -- Executing [80620@huji-remote-gr:12] MYSQL("SIP/80611-b7d56a88", "Fetch FetchId 2 CalledName") in new stack  -- Executing [80620@huji-remote-gr:13] MYSQL("SIP/80611-b7d56a88", "Clear 2") in new stack  -- Executing [80620@huji-remote-gr:14] GotoIf("SIP/80611-b7d56a88", "0?15:21") in new stack  -- Goto (huji-remote-gr,80620,21)  -- Executing [80620@huji-remote-gr:21] NoOp("SIP/80611-b7d56a88", "Finish if-huji-local-7") in new stack  -- Executing [80620@huji-remote-gr:22] MYSQL("SIP/80611-b7d56a88", "Disconnect 1") in new stack  -- Executing [80620@huji-remote-gr:23] Dial("SIP/80611-b7d56a88", "SIP/80620,20,L(3600000:60000:30000)") in new stack  -- Limit Data for this call:  > timelimit = 3600000  > play_warning = 60000  > play_to_caller = yes  > play_to_callee = no  > warning_freq = 30000  > start_sound =  > warning_sound = timeleft  > end_sound =  == Using TOS bits 0  == Using CoS mark 5  == Using TOS bits 0  == Using CoS mark 6  == Using TOS bits 0  == Using CoS mark 5 Audio is at 132.64.9.164 port 2636 failed to extend from 0 to 44 failed to extend from 0 to 171 failed to extend from 0 to 69 Adding codec 0x8 (alaw) to SDP failed to extend from 0 to 43 failed to extend from 0 to 48 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 132.64.4.137:2135: NOTIFY sip:85684@132.64.4.137:2135;line=mpqvu5xe SIP/2.0 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK7f97dbd4;rport Max-Forwards: 70 From: ;tag=as2b663d6f To: ;tag=w0gasmyayd Contact: Call-ID: 3c26700a6ddd-7f3p477qk4w1@snom320-0004132480B4 CSeq: 103 NOTIFY User-Agent: Asterisk PBX SVN-trunk-r83213M Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 0 ---  == Extension Changed 80620 new state Ringing for Notify User 85684 Reliably Transmitting (no NAT) to 132.64.4.44:5060: INVITE sip:80620@132.64.4.44 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK75ae2389;rport Max-Forwards: 70 From: "Yehavi" ;tag=as576919fa To: Contact: Call-ID: 574acdf152f67961218adb2a746df850@132.64.9.164 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-trunk-r83213M Remote-Party-ID: "Yehavi" ;privacy=off;screen=no Date: Thu, 20 Sep 2007 13:51:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 272 v=0 o=root 494654327 494654327 IN IP4 132.64.4.120 s=session c=IN IP4 132.64.4.120 t=0 0 m=audio 16384 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv ---  -- Called 80620 <--- SIP read from 132.64.4.44:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK75ae2389;rport From: "Yehavi" ;tag=as576919fa To: ;tag=1c506961039 Call-ID: 574acdf152f67961218adb2a746df850@132.64.9.164 CSeq: 102 INVITE Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.60A.016.003 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 132.64.4.137:2135 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK7f97dbd4;rport=5060 From: ;tag=as2b663d6f To: ;tag=w0gasmyayd Call-ID: 3c26700a6ddd-7f3p477qk4w1@snom320-0004132480B4 CSeq: 103 NOTIFY Content-Length: 0 <-------------> --- (7 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 132.64.4.44:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK75ae2389;rport From: "Yehavi" ;tag=as576919fa To: ;tag=1c506961039 Call-ID: 574acdf152f67961218adb2a746df850@132.64.9.164 CSeq: 102 INVITE Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.60A.016.003 Content-Length: 0 <-------------> --- (11 headers 0 lines) ---  -- SIP/80620-0883d148 is ringing <--- Transmitting (no NAT) to 132.64.4.120:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 132.64.4.120:5060;branch=z9hG4bKa61b0b03ee7aad0d;received=132.64.4.120 From: line2 ATAtest ;tag=951678558 To: ;tag=as3b305c04 Call-ID: 277848150@132.64.4.120 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r83213M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- SIP read from 132.64.4.44:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK75ae2389;rport From: "Yehavi" ;tag=as576919fa To: ;tag=1c506961039 Call-ID: 574acdf152f67961218adb2a746df850@132.64.9.164 CSeq: 102 INVITE Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.60A.016.003 Content-Type: application/sdp Content-Length: 229 v=0 o=AudiocodesGW 506969384 506969303 IN IP4 132.64.4.44 s=Phone-Call c=IN IP4 132.64.4.44 t=0 0 m=audio 6000 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (12 headers 11 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 132.64.4.44:6000 Found description format pcma for ID 8 Found description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 132.64.4.44:6000 --- set_address_from_contact host '132.64.4.44' list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.44, port 5060 failed to extend from 0 to 44 failed to extend from 0 to 171 Transmitting (no NAT) to 132.64.4.44:5060: ACK sip:80620@132.64.4.44 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK59e20384;rport Max-Forwards: 70 From: "Yehavi" ;tag=as576919fa To: ;tag=1c506961039 Contact: Call-ID: 574acdf152f67961218adb2a746df850@132.64.9.164 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-trunk-r83213M Remote-Party-ID: "Yehavi" ;privacy=off;screen=no Content-Length: 0 --- failed to extend from 0 to 42 failed to extend from 0 to 48 failed to extend from 0 to 48 Reliably Transmitting (no NAT) to 132.64.4.137:2135: NOTIFY sip:85684@132.64.4.137:2135;line=mpqvu5xe SIP/2.0 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK123b5404;rport Max-Forwards: 70 From: ;tag=as2b663d6f To: ;tag=w0gasmyayd Contact: Call-ID: 3c26700a6ddd-7f3p477qk4w1@snom320-0004132480B4 CSeq: 104 NOTIFY User-Agent: Asterisk PBX SVN-trunk-r83213M Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 0 ---  == Extension Changed 80620 new state InUse for Notify User 85684  -- SIP/80620-0883d148 answered SIP/80611-b7d56a88 Audio is at 132.64.9.164 port 34864 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 132.64.4.120:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.4.120:5060;branch=z9hG4bKa61b0b03ee7aad0d;received=132.64.4.120 From: line2 ATAtest ;tag=951678558 To: ;tag=as3b305c04 Call-ID: 277848150@132.64.4.120 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r83213M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 247 v=0 o=root 1646592573 1646592573 IN IP4 132.64.4.44 s=session c=IN IP4 132.64.4.44 t=0 0 m=audio 6000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> <--- SIP read from 132.64.4.120:5060 ---> ACK sip:80620@132.64.9.164 SIP/2.0 Via: SIP/2.0/UDP 132.64.4.120:5060;branch=z9hG4bK9b326d12ff71668e From: line2 ATAtest ;tag=951678558 To: ;tag=as3b305c04 Call-ID: 277848150@132.64.4.120 CSeq: 2 ACK User-Agent: Cisco ATA 186 v3.2.0 atasip (041111A) Authorization: Digest username="80611",realm="cc.huji.ac.il",nonce="528091f4",uri="sip:80620@pbx-dev.cc.huji.ac.il",response="4db55781d7c1d04a71db8510e1d5d010" Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from 132.64.4.137:2135 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK123b5404;rport=5060 From: ;tag=as2b663d6f To: ;tag=w0gasmyayd Call-ID: 3c26700a6ddd-7f3p477qk4w1@snom320-0004132480B4 CSeq: 104 NOTIFY Content-Length: 0 <-------------> --- (7 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived Really destroying SIP dialog '3c26700a3d09-0gsczhjqwmf1@snom320-0004132480B4' Method: SUBSCRIBE Really destroying SIP dialog '3c26700a3d09-vj6y9ke9ouf1@snom320-0004132480B4' Method: SUBSCRIBE Really destroying SIP dialog '3c26700a3a98-7f3p477qk4w1@snom320-0004132480B4' Method: SUBSCRIBE Retransmitting #6 (NAT) to 132.64.4.137:2135: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 132.64.4.137:2135;branch=z9hG4bK-3vhk6hianhcb;received=132.64.4.137;rport=2135 From: ;tag=cypxxhdq5i To: ;tag=as79f54839 Call-ID: 3c26700a445c-wjx5tt2ysc3u@snom320-0004132480B4 CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX SVN-trunk-r83213M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Really destroying SIP dialog '3c26700a186a-sbiow7lwpu03@snom320-0004132480B4' Method: REGISTER <--- SIP read from 132.64.4.44:5060 ---> BYE sip:80611@132.64.9.164 SIP/2.0 Via: SIP/2.0/UDP 132.64.4.44;branch=z9hG4bKac515474443 Max-Forwards: 70 From: ;tag=1c506961039 To: "Yehavi" ;tag=as576919fa Call-ID: 574acdf152f67961218adb2a746df850@132.64.9.164 CSeq: 1 BYE Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.60A.016.003 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 132.64.4.44 : 5060 (no NAT) <--- Transmitting (no NAT) to 132.64.4.44:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.4.44;branch=z9hG4bKac515474443;received=132.64.4.44 From: ;tag=1c506961039 To: "Yehavi" ;tag=as576919fa Call-ID: 574acdf152f67961218adb2a746df850@132.64.9.164 CSeq: 1 BYE User-Agent: Asterisk PBX SVN-trunk-r83213M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------>  == Spawn extension (huji-remote-gr, 80620, 23) exited non-zero on 'SIP/80611-b7d56a88'  -- Executing [h@huji-remote-gr:1] ResetCDR("SIP/80611-b7d56a88", "w") in new stack failed to extend from 0 to 44 failed to extend from 0 to 171 failed to extend from 0 to 42 failed to extend from 0 to 49 failed to extend from 0 to 48 Reliably Transmitting (no NAT) to 132.64.4.137:2135: NOTIFY sip:85684@132.64.4.137:2135;line=mpqvu5xe SIP/2.0 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK0be3387e;rport Max-Forwards: 70 From: ;tag=as2b663d6f To: ;tag=w0gasmyayd Contact: Call-ID: 3c26700a6ddd-7f3p477qk4w1@snom320-0004132480B4 CSeq: 105 NOTIFY User-Agent: Asterisk PBX SVN-trunk-r83213M Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 0 ---  == Extension Changed 80620 new state Idle for Notify User 85684  -- Executing [h@huji-remote-gr:2] NoOp("SIP/80611-b7d56a88", "80611") in new stack  -- Executing [h@huji-remote-gr:3] Set("SIP/80611-b7d56a88", "tmp=") in new stack  -- Executing [h@huji-remote-gr:4] GotoIf("SIP/80611-b7d56a88", "0?5:11") in new stack  -- Goto (huji-remote-gr,h,11)  -- Executing [h@huji-remote-gr:11] NoOp("SIP/80611-b7d56a88", "Finish if-huji-local-12") in new stack  -- Executing [h@huji-remote-gr:12] Set("SIP/80611-b7d56a88", "tmp=") in new stack  -- Executing [h@huji-remote-gr:13] GotoIf("SIP/80611-b7d56a88", "0?14:20") in new stack  -- Goto (huji-remote-gr,h,20)  -- Executing [h@huji-remote-gr:20] NoOp("SIP/80611-b7d56a88", "Finish if-huji-local-13") in new stack Scheduling destruction of SIP dialog '277848150@132.64.4.120' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.120, port 5060 Reliably Transmitting (no NAT) to 132.64.4.120:5060: BYE sip:80611@132.64.4.120:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK1bb3c412;rport Max-Forwards: 70 From: ;tag=as3b305c04 To: line2 ATAtest ;tag=951678558 Call-ID: 277848150@132.64.4.120 CSeq: 102 BYE User-Agent: Asterisk PBX SVN-trunk-r83213M Content-Length: 0 --- <--- SIP read from 132.64.4.120:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK1bb3c412;rport From: ;tag=as3b305c04 To: line2 ATAtest ;tag=951678558 Call-ID: 277848150@132.64.4.120 CSeq: 102 BYE Server: Cisco ATA 186 v3.2.0 atasip (041111A) Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '574acdf152f67961218adb2a746df850@132.64.9.164' Method: BYE Really destroying SIP dialog '277848150@132.64.4.120' Method: ACK <--- SIP read from 132.64.4.137:2135 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK0be3387e;rport=5060 From: ;tag=as2b663d6f To: ;tag=w0gasmyayd Call-ID: 3c26700a6ddd-7f3p477qk4w1@snom320-0004132480B4 CSeq: 105 NOTIFY Content-Length: 0 <-------------> --- (7 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived [Sep 20 15:51:43] WARNING[8200]: chan_sip.c:2299 retrans_pkt: Maximum retries exceeded on transmission 3c26700a445c-wjx5tt2ysc3u@snom320-0004132480B4 for seqno 1 (Critical Response) Really destroying SIP dialog '3c26700a445c-wjx5tt2ysc3u@snom320-0004132480B4' Method: SUBSCRIBE stop now Beginning asterisk shutdown.... Executing last minute cleanups  == Destroying musiconhold processes Asterisk cleanly ending (0).