Asterisk SVN-branch-1.4-r115017, Copyright (C) 1999 - 2008 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. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': Found  == Parsing '/etc/asterisk/extconfig.conf': Found  == 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  == Manager registered action Ping  == Manager registered action Events  == Manager registered action Logoff  == Manager registered action Hangup  == Manager registered action Status  == Manager registered action Setvar  == Manager registered action Getvar  == Manager registered action GetConfig  == 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 UserEvent  == Manager registered action WaitEvent  == Parsing '/etc/asterisk/manager.conf': Found  == Parsing '/etc/asterisk/cdr.conf': Found [May 1 18:24:04] NOTICE[3973]: cdr.c:1366 do_reload: CDR simple logging enabled.  == Parsing '/etc/asterisk/rtp.conf': Found  == RTP Allocating from port range 5000 -> 32767  == Parsing '/etc/asterisk/udptl.conf': Found  == UDPTL allocating from port range 1500 -> 4999 Asterisk PBX Core Initializing Registering builtin applications:  [Answer]  == Registered application 'Answer'  [BackGround]  == Registered application 'BackGround'  [Busy]  == Registered application 'Busy'  [Congestion]  == Registered application 'Congestion'  [Goto]  == Registered application 'Goto'  [GotoIf]  == Registered application 'GotoIf'  [GotoIfTime]  == Registered application 'GotoIfTime'  [ExecIfTime]  == Registered application 'ExecIfTime'  [Hangup]  == Registered application 'Hangup'  [NoOp]  == Registered application 'NoOp'  [Progress]  == Registered application 'Progress'  [ResetCDR]  == Registered application 'ResetCDR'  [Ringing]  == Registered application 'Ringing'  [SayNumber]  == Registered application 'SayNumber'  [SayDigits]  == Registered application 'SayDigits'  [SayAlpha]  == Registered application 'SayAlpha'  [SayPhonetic]  == Registered application 'SayPhonetic'  [SetAMAFlags]  == Registered application 'SetAMAFlags'  [SetGlobalVar]  == Registered application 'SetGlobalVar'  [Set]  == Registered application 'Set'  [ImportVar]  == Registered application 'ImportVar'  [Wait]  == Registered application 'Wait'  [WaitExten]  == Registered application 'WaitExten'  == Manager registered action DBGet  == Manager registered action DBPut  == Parsing '/etc/asterisk/enum.conf': Found Asterisk Dynamic Loader Starting:  == Parsing '/etc/asterisk/modules.conf': Found [May 1 18:24:04] NOTICE[3973]: loader.c:851 load_modules: 134 modules will be loaded.  == Parsing '/etc/asterisk/indications.conf': Found  -- Registered indication country 'cl'  -- Registered indication country 'tw'  -- Registered indication country 'us'  -- Registered indication country 'au'  -- Registered indication country 'fr'  -- Registered indication country 'de'  -- Registered indication country 'nl'  -- Registered indication country 'uk'  -- Registered indication country 'fi'  -- Registered indication country 'no'  -- Registered indication country 'br'  -- Registered indication country 'za'  -- Registered indication country 'it'  -- Registered indication country 'us-o'  -- Registered indication country 'gr'  -- Registered indication country 'ru'  -- Registered indication country 'nz'  -- Setting default indication country to 'us'  == Registered application 'PlayTones'  == Registered application 'StopPlayTones' res_indications.so => (Indications Resource)  == Parsing '/etc/asterisk/features.conf': Found  -- Registered extension context 'parkedcalls'  -- Added extension '700' priority 1 to parkedcalls  == Registered application 'ParkedCall'  == Registered application 'Park'  == Manager registered action ParkedCalls  == Manager registered action Park res_features.so => (Call Features Resource)  == 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) [May 1 18:24:04] NOTICE[3973]: config.c:1270 ast_config_engine_register: Registered Config Engine odbc res_config_odbc loaded. res_config_odbc.so => (ODBC Configuration)  == Registered custom function SMDI_MSG_RETRIEVE  == Registered custom function SMDI_MSG [May 1 18:24:04] NOTICE[3973]: res_smdi.c:814 smdi_load: Unable to load config smdi.conf: SMDI disabled [May 1 18:24:04] WARNING[3973]: res_smdi.c:1264 load_module: No SMDI interfaces are available to listen on, not starting SMDI listener. res_speech.so => (Generic Speech Recognition API)  == Registered application 'DeadAGI'  == Registered application 'EAGI'  == Registered application 'AGI' res_agi.so => (Asterisk Gateway Interface (AGI))  == Parsing '/etc/asterisk/adsi.conf': Found res_adsi.so => (ADSI Resource)  -- Loaded PUBLIC key 'iaxtel'  -- Loaded PUBLIC key 'freeworlddialup' res_crypto.so => (Cryptographic Digital Signatures) res_clioriginate.so => (Call origination from the CLI)  == Registered file format h263, extension(s) h263 format_h263.so => (Raw H.263 data)  == Registered custom function IAXPEER  == Registered application 'IAX2Provision'  == Manager registered action IAXpeers  == Manager registered action IAXnetstats  == Parsing '/etc/asterisk/iax.conf': Found [May 1 18:24:04] WARNING[3973]: acl.c:475 ast_str2tos: TOS value lowdelay is deprecated. Please see doc/ip-tos.txt for more information. [May 1 18:24:04] WARNING[3973]: acl.c:475 ast_str2tos: TOS value lowdelay is deprecated. Please see doc/ip-tos.txt for more information.  == Using TOS bits 16  == Binding IAX2 to default address 0.0.0.0:4569  > doing dnsmgr_lookup for '216.207.245.47'  == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2))  == 10 helper threaads started  == IAX Ready and Listening  == Loaded firmware 'iaxy.bin'  == Parsing '/etc/asterisk/iaxprov.conf': Found [May 1 18:24:04] WARNING[3973]: acl.c:475 ast_str2tos: TOS value lowdelay is deprecated. Please see doc/ip-tos.txt for more information.  -- Loaded provisioning template 'default' chan_iax2.so => (Inter Asterisk eXchange (Ver 2))  == Parsing '/etc/asterisk/codecs.conf': Found  == Registered translator 'alawtolin' from format alaw to slin, cost 1  == Registered translator 'lintoalaw' from format slin to alaw, cost 1 codec_alaw.so => (A-law Coder/Decoder)  == Registered application 'Log'  == Registered application 'Verbose' app_verbose.so => (Send verbose output)  == Registered file format g723sf, extension(s) g723|g723sf format_g723.so => (G.723.1 Simple Timestamp File Format)  == Registered application 'SetCallerPres'  == Registered application 'SetCallerID' app_setcallerid.so => (Set CallerID Application)  == Registered custom function ENV  == Registered custom function STAT func_env.so => (Environment/filesystem dialplan functions)  == Registered application 'ControlPlayback' app_controlplayback.so => (Control Playback Application)  == Parsing '/etc/asterisk/cdr.conf': Found  == Registered custom function RAND func_rand.so => (Random number dialplan function)  == Parsing '/etc/asterisk/queues.conf': Found  == 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 QueueAdd  == Manager registered action QueueRemove  == Manager registered action QueuePause  == Registered custom function QUEUEAGENTCOUNT  == 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 'ExternalIVR' app_externalivr.so => (External IVR Interface Application)  == Parsing '/etc/asterisk/cdr_manager.conf': Found cdr_manager.so => (Asterisk Manager Interface CDR Backend)  == Registered application 'SendURL' app_url.so => (Send URL Applications)  == Registered custom function MUSICCLASS func_moh.so => (Music-on-hold dialplan function)  == Registered application 'Milliwatt' app_milliwatt.so => (Digital Milliwatt (mu-law) Test Application)  == 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 GROUP_COUNT  == Registered custom function GROUP_MATCH_COUNT  == Registered custom function GROUP_LIST  == Registered custom function GROUP func_groupcount.so => (Channel group dialplan functions)  == Registered application 'ICES' app_ices.so => (Encode and Stream via icecast and ices)  == Registered custom function REALTIME func_realtime.so => (Read/Write values from a RealTime repository)  == Registered custom function BLACKLIST  == Registered application 'LookupBlacklist' app_lookupblacklist.so => (Look up Caller*ID name/number from blacklist database)  == Registered application 'Morsecode' app_morsecode.so => (Morse code)  == Parsing '/etc/asterisk/codecs.conf': Found  == Registered translator 'g726tolin' from format g726 to slin, cost 1  == Registered translator 'lintog726' from format slin to g726, cost 1  == Registered translator 'g726aal2tolin' from format g726aal2 to slin, cost 1  == Registered translator 'lintog726aal2' from format slin to g726aal2, cost 1  == Registered translator 'g726aal2tog726' from format g726aal2 to g726, cost 1  == Registered translator 'g726tog726aal2' from format g726 to g726aal2, cost 1 codec_g726.so => (ITU G.726-32kbps G726 Transcoder)  == Registered application 'WaitForRing' app_waitforring.so => (Waits until first ring after time)  == Registered file format g729, extension(s) g729 format_g729.so => (Raw G729 data)  == Registered application 'ChannelRedirect' app_channelredirect.so => (Channel Redirect)  == Registered channel type 'Agent' (Call Agent Proxy Channel)  == Parsing '/etc/asterisk/agents.conf': Found  == Registered application 'AgentLogin'  == Registered application 'AgentCallbackLogin'  == Registered application 'AgentMonitorOutgoing'  == Manager registered action Agents  == Manager registered action AgentLogoff  == Manager registered action AgentCallbackLogin  == Registered custom function AGENT chan_agent.so => (Agent Proxy Channel)  == Parsing '/etc/asterisk/codecs.conf': Found  == Registered translator 'lpc10tolin' from format lpc10 to slin, cost 1  == Registered translator 'lintolpc10' from format slin to lpc10, cost 2 codec_lpc10.so => (LPC10 2.4kbps Coder/Decoder) res_convert.so => (File format conversion CLI command)  == Registered application 'LookupCIDName' app_lookupcidname.so => (Look up CallerID Name from local database)  == Parsing '/etc/asterisk/skinny.conf': Found [May 1 18:24:04] WARNING[3973]: chan_skinny.c:4760 reload_config: Option 'port' at line 5 of skinny.conf has been deprecated. Please use 'bindport' instead.  == Skinny listening on 0.0.0.0:2000  == Registered channel type 'Skinny' (Skinny Client Control Protocol (Skinny)) pbx_loopback.so => (Loopback Switch)  == Registered translator 'alawtoulaw' from format alaw to ulaw, cost 1  == Registered translator 'ulawtoalaw' from format ulaw to alaw, cost 1 codec_a_mu.so => (A-law and Mulaw direct Coder/Decoder) pbx_spool.so => (Outgoing Spool Support)  == Registered application 'TrySystem'  == Registered application 'System' app_system.so => (Generic System() application)  == Registered application 'Exec'  == Registered application 'TryExec'  == Registered application 'ExecIf' app_exec.so => (Executes dialplan applications) pbx_realtime.so => (Realtime Switch)  == Parsing '/etc/asterisk/oss.conf': Found  == Registered channel type 'Console' (OSS Console Channel Driver) chan_oss.so => (OSS Console Channel Driver)  == Registered custom function MATH func_math.so => (Mathematical dialplan function)  == Registered application 'WaitForSilence' app_waitforsilence.so => (Wait For Silence)  == Registered file format wav49, extension(s) WAV|wav49 format_wav_gsm.so => (Microsoft WAV format (Proprietary GSM))  == Parsing '/etc/asterisk/codecs.conf': Found  == Registered translator 'adpcmtolin' from format adpcm to slin, cost 1  == Registered translator 'lintoadpcm' from format slin to adpcm, cost 1 codec_adpcm.so => (Adaptive Differential PCM Coder/Decoder)  == 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 'DumpChan' app_dumpchan.so => (Dump Info About The Calling Channel)  == Registered application 'ChanIsAvail' app_chanisavail.so => (Check channel availability)  == Parsing '/etc/asterisk/sip.conf': Found  == SIP Listening on 0.0.0.0:5060  == Using SIP TOS: none  == 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 application 'GetCPEID' app_getcpeid.so => (Get ADSI CPE ID)  == Registered custom function CUT  == Registered custom function SORT func_cut.so => (Cut out information from a string)  == Registered application 'TestClient'  == Registered application 'TestServer' app_test.so => (Interface Test Application)  == Registered application 'ReadFile' app_readfile.so => (Stores output of file into a variable)  == Registered custom function DB  == Registered custom function DB_EXISTS  == Registered custom function DB_DELETE func_db.so => (Database (astdb) related dialplan functions)  == Registered application 'StackPop'  == Registered application 'Return'  == Registered application 'GosubIf'  == Registered application 'Gosub' app_stack.so => (Stack Routines)  == Registered application 'SendText' app_sendtext.so => (Send Text Applications)  == Registered application 'BackgroundDetect' app_talkdetect.so => (Playback with Talk Detection)  == Registered application 'ChanSpy'  == Registered application 'ExtenSpy' app_chanspy.so => (Listen to the audio of an active channel)  == Registered custom function MD5  == Registered custom function CHECK_MD5 func_md5.so => (MD5 digest dialplan functions)  == Registered custom function VMCOUNT  == Registered application 'HasVoicemail'  == Registered application 'HasNewVoicemail' app_hasnewvoicemail.so => (Indicator for whether a voice mailbox has messages in a given folder.)  == Registered application 'NBScat' app_nbscat.so => (Silly NBS Stream Application)  == Registered custom function URIDECODE  == Registered custom function URIENCODE func_uri.so => (URI encode/decode dialplan functions)  == Registered application 'MixMonitor'  == Registered application 'StopMixMonitor' app_mixmonitor.so => (Mixed Audio Monitoring Application)  == Registered application 'SoftHangup' app_softhangup.so => (Hangs up the requested channel)  == Registered application 'MP3Player' app_mp3.so => (Silly MP3 Application)  == Registered application 'SetCDRUserField'  == Registered application 'AppendCDRUserField'  == Manager registered action SetCDRUserField app_setcdruserfield.so => (CDR user field apps)  == Registered custom function ISNULL  == Registered custom function SET  == Registered custom function EXISTS  == Registered custom function IF  == Registered custom function IFTIME func_logic.so => (Logical dialplan functions)  == Registered application 'SMS' app_sms.so => (SMS/PSTN handler)  == Registered custom function ENUMLOOKUP  == Registered custom function TXTCIDNAME func_enum.so => (ENUM related dialplan functions)  == Registered application 'RealTimeUpdate'  == Registered application 'RealTime' app_realtime.so => (Realtime Data Lookup/Rewrite)  == Registered application 'DBdel'  == Registered application 'DBdeltree' app_db.so => (Database Access Functions)  == Registered application 'NoCDR' app_cdr.so => (Tell Asterisk to not maintain a CDR for the current call)  == Registered application 'MacroExit'  == Registered application 'MacroIf'  == Registered application 'MacroExclusive'  == Registered application 'Macro' app_macro.so => (Extension Macros)  == Registered custom function CURL func_curl.so => (Load external URL)  == Parsing '/etc/asterisk/festival.conf': Found  == Registered application 'Festival' app_festival.so => (Simple Festival Interface)  == Registered application 'Dictate' app_dictate.so => (Virtual Dictation Machine)  == Registered application 'Authenticate' app_authenticate.so => (Authentication Application)  == Registered custom function CALLERID func_callerid.so => (Caller ID related dialplan function) [May 1 18:24:04] WARNING[3973]: app_followme.c:303 reload_followme: No follow me config file (followme.conf), so no follow me  == Registered application 'VoiceMail'  == Registered application 'VoiceMailMain'  == Registered application 'MailboxExists'  == Registered application 'VMAuthenticate'  == Parsing '/etc/asterisk/voicemail.conf': Found app_voicemail.so => (Comedian Mail (Voicemail System))  == Registered application 'ForkCDR' app_forkcdr.so => (Fork The CDR into 2 separate entities)  == Registered custom function GLOBAL func_global.so => (Global variable dialplan functions)  == Registered application 'UserEvent' app_userevent.so => (Custom User Event Application)  == Registered custom function TIMEOUT func_timeout.so => (Channel timeout dialplan functions)  == Registered application 'SayUnixTime'  == Registered application 'DateTime' app_sayunixtime.so => (Say time)  == Registered application 'Record' app_record.so => (Trivial Record Application)  == Registered file format iLBC, extension(s) ilbc format_ilbc.so => (Raw iLBC data)  == Parsing '/etc/asterisk/codecs.conf': Found  == Registered translator 'ulawtolin' from format ulaw to slin, cost 1  == Registered translator 'lintoulaw' from format slin to ulaw, cost 1 codec_ulaw.so => (mu-Law Coder/Decoder)  == Registered custom function CHANNEL func_channel.so => (Channel information dialplan function)  == Registered application 'Read' app_read.so => (Read Variable Application)  == Registered application 'ParkAndAnnounce' app_parkandannounce.so => (Call Parking and Announce Application)  == Registered file format gsm, extension(s) gsm format_gsm.so => (Raw GSM data)  == Registered custom function CDR func_cdr.so => (CDR dialplan function)  == Registered application 'PrivacyManager' app_privacy.so => (Require phone number to be entered, if no CallerID sent)  == Registered channel type 'Local' (Local Proxy Channel Driver) chan_local.so => (Local Proxy Channel (Note: used internally by other modules))  == Registered application 'Playback' app_playback.so => (Sound File Playback Application)  == Manager registered action PlayDTMF  == Registered application 'SendDTMF' app_senddtmf.so => (Send DTMF digits Application)  == Registered application 'Zapateller' app_zapateller.so => (Block Telemarketers with Special Information Tone)  == Parsing '/etc/asterisk/h323.conf': Found  == Creating H.323 Endpoint  == Setting default context to default  == Registered channel type 'H323' (The NuFone Network's Open H.323 Channel Driver)  == H.323 listener started chan_h323.so => (The NuFone Network's OpenH323 Channel Driver)  == Registered application 'SetTransferCapability' app_settransfercapability.so => (Set ISDN Transfer Capability)  == Registered custom function SHA1 func_sha1.so => (SHA-1 computation 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 func_strings.so => (String handling dialplan functions)  == Registered application 'DISA' app_disa.so => (DISA (Direct Inward System Access) Application)  == Registered custom function BASE64_ENCODE  == Registered custom function BASE64_DECODE func_base64.so => (base64 encode/decode dialplan functions)  == Parsing '/etc/asterisk/alarmreceiver.conf': Found  == Registered application 'AlarmReceiver' app_alarmreceiver.so => (Alarm Receiver for Asterisk)  == Registered application 'Transfer' app_transfer.so => (Transfer)  == 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 file format h264, extension(s) h264 format_h264.so => (Raw H.264 data)  == Parsing '/etc/asterisk/phone.conf': Found  == Registered channel type 'Phone' (Standard Linux Telephony API Driver) chan_phone.so => (Linux Telephony API Support)  == Registered application 'Directory' app_directory.so => (Extension Directory)  == Registered file format vox, extension(s) vox format_vox.so => (Dialogic VOX (ADPCM) File Format)  == Registered file format wav, extension(s) wav format_wav.so => (Microsoft WAV format (8000Hz Signed Linear))  == Parsing '/etc/asterisk/codecs.conf': Found  == Registered translator 'gsmtolin' from format gsm to slin, cost 1  == Registered translator 'lintogsm' from format slin to gsm, cost 1 codec_gsm.so => (GSM Coder/Decoder)  == Registered application 'ADSIProg' app_adsiprog.so => (Asterisk ADSI Programming Application)  == Parsing '/etc/asterisk/extensions.conf': Found  -- Registered extension context 'default'  -- Added extension '_X.' priority 1 to default  -- Added extension '_X.' priority 2 to default  -- Added extension '_X.' priority 3 to default  -- Added extension '_X.' priority 4 to default  -- Added extension '_X.' priority 5 to default  -- Added extension '_X.' priority 6 to default  -- Added extension '_X.' priority 7 to default  -- Added extension '_X.' priority 8 to default  -- Added extension '_X.' priority 9 to default  -- Added extension '_X.' priority 10 to default  -- Added extension '_X.' priority 100 to default  -- Added extension '_X.' priority 300 to default  -- Added extension '_X.' priority 301 to default  -- Added extension '_X.' priority 302 to default  -- Registered extension context 'default204'  -- Added extension '_X.' priority 1 to default204  -- Added extension '_X.' priority 2 to default204  -- Added extension '_X.' priority 3 to default204  -- Added extension '_X.' priority 4 to default204  -- Added extension '_X.' priority 5 to default204  -- Added extension '_X.' priority 6 to default204  -- Added extension '_X.' priority 7 to default204  -- Added extension '_X.' priority 8 to default204  -- Added extension '_X.' priority 9 to default204  -- Added extension '_X.' priority 10 to default204  -- Added extension '_X.' priority 11 to default204  -- Added extension '_X.' priority 12 to default204  -- Added extension 'closecall' priority 1 to default204  -- Added extension 'closecall' priority 2 to default204  -- Registered extension context 'sendmail'  -- Added extension '_X.' priority 1 to sendmail  -- Added extension '_X.' priority 2 to sendmail  -- Added extension 'h' priority 1 to sendmail  -- Added extension 'h' priority 2 to sendmail  -- Registered extension context 'process'  -- Added extension '_X.' priority 1 to process  -- Added extension '_X.' priority 2 to process  -- Added extension '_X.' priority 3 to process  -- Added extension '_X.' priority 4 to process  -- Added extension '_X.' priority 5 to process  -- Added extension '_X.' priority 6 to process  -- Added extension '_X.' priority 7 to process  -- Added extension '_X.' priority 8 to process  -- Added extension '_X.' priority 9 to process  -- Added extension '_X.' priority 100 to process  -- Added extension '_X.' priority 300 to process  -- Added extension '_X.' priority 301 to process  -- Added extension '_X.' priority 302 to process pbx_config.so => (Text Extension Configuration)  == Registered format 'jpg' (JPEG (Joint Picture Experts Group)) format_jpeg.so => (JPEG (Joint Picture Experts Group) Image Format)  == Registered custom function LANGUAGE func_language.so => (Channel language dialplan function) [May 1 18:24:04] WARNING[3973]: cdr_custom.c:99 load_config: Failed to load configuration file. Module not activated. [May 1 18:24:04] NOTICE[3973]: pbx_ael.c:4137 pbx_load_module: Starting AEL load process. [May 1 18:24:04] NOTICE[3973]: pbx_ael.c:4144 pbx_load_module: AEL load process: calculated config file name '/etc/asterisk/extensions.ael'. [May 1 18:24:04] NOTICE[3973]: pbx_ael.c:4147 pbx_load_module: File /etc/asterisk/extensions.ael not found; AEL declining load  == Registered application 'Pickup' app_directed_pickup.so => (Directed Call Pickup Application)  == Registered application 'SendImage' app_image.so => (Image Transmission Application)  == Registered file format sln, extension(s) sln|raw format_sln.so => (Raw Signed Linear Audio support (SLN))  == Registered application 'Dial'  == Registered application 'RetryDial' app_dial.so => (Dialing Application)  == Registered application 'While'  == Registered application 'EndWhile'  == Registered application 'ExitWhile'  == Registered application 'ContinueWhile' app_while.so => (While Loops and Conditional Execution)  == Registered application 'Echo' app_echo.so => (Simple Echo Application)  == Parsing '/etc/asterisk/mgcp.conf': Found  == MGCP Listening on 0.0.0.0:2727  == Using TOS bits 0  == Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP)) chan_mgcp.so => (Media Gateway Control Protocol (MGCP))  == Registered application 'Random' app_random.so => (Random goto) Asterisk Ready. *CLI> [May 1 18:24:04] NOTICE[3997]: chan_sip.c:12656 handle_response_peerpoke: Peer '7864335989' is now Reachable. (86ms / 8000ms) [May 1 18:24:04] NOTICE[3997]: chan_sip.c:12656 handle_response_peerpoke: Peer 'sipserver' is now Reachable. (9ms / 8000ms) [May 1 18:24:04] NOTICE[3997]: chan_sip.c:12656 handle_response_peerpoke: Peer 'sipserver204' is now Reachable. (1ms / 8000ms) *CLI> *CLI> sip set debug SIP Debugging enabled *CLI> <--- SIP read from 72.187.243.97:45990 ---> REGISTER sip:66.28.190.219 SIP/2.0 Via: SIP/2.0/UDP 72.187.243.97:45990;branch=z9hG4bK0B023A7FE14A48AEBE189C2EE376E0A9 From: 7864335989 ;tag=1867072463 To: 7864335989 Contact: "7864335989" Call-ID: FF611B51FF2F4A3B9715B926FB8E7F7A@66.28.190.219 CSeq: 65629 REGISTER Expires: 60 Max-Forwards: 70 User-Agent: X-PRO release 1105x Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 72.187.243.97 : 45990 (NAT) <--- Transmitting (NAT) to 72.187.243.97:45990 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 72.187.243.97:45990;branch=z9hG4bK0B023A7FE14A48AEBE189C2EE376E0A9;received=72.187.243.97 From: 7864335989 ;tag=1867072463 To: 7864335989 Call-ID: FF611B51FF2F4A3B9715B926FB8E7F7A@66.28.190.219 CSeq: 65629 REGISTER User-Agent: Cisco 3845 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------>  -- Saved useragent "X-PRO release 1105x" for peer 7864335989 <--- Transmitting (NAT) to 72.187.243.97:45990 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 72.187.243.97:45990;branch=z9hG4bK0B023A7FE14A48AEBE189C2EE376E0A9;received=72.187.243.97 From: 7864335989 ;tag=1867072463 To: 7864335989 ;tag=as104f6699 Call-ID: FF611B51FF2F4A3B9715B926FB8E7F7A@66.28.190.219 CSeq: 65629 REGISTER User-Agent: Cisco 3845 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Thu, 01 May 2008 22:24:10 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'FF611B51FF2F4A3B9715B926FB8E7F7A@66.28.190.219' in 32000 ms (Method: REGISTER) <--- SIP read from 72.187.243.97:45990 ---> INVITE sip:14166289921@66.28.190.219 SIP/2.0 Via: SIP/2.0/UDP 72.187.243.97:45990;branch=z9hG4bK137F96CF17EE4DC2AF466F1A35F17BEB From: 7864335989 ;tag=168603557 To: Contact: Call-ID: E0C19ACC-FCCB-4193-8907-6FA09F3A3870@192.168.1.111 CSeq: 12134 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-PRO release 1105x Content-Length: 217 v=0 o=7864335989 144387732 144387830 IN IP4 72.187.243.97 s=X-PRO c=IN IP4 72.187.243.97 t=0 0 m=audio 45992 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> --- (11 headers 10 lines) --- Sending to 72.187.243.97 : 45990 (NAT) Using INVITE request as basis request - E0C19ACC-FCCB-4193-8907-6FA09F3A3870@192.168.1.111 Found peer '7864335989' Found RTP audio format 18 Found RTP audio format 101 [May 1 18:24:13] DEBUG[3997]: chan_sip.c:5208 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 72.187.243.97:45992 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 72.187.243.97:45992 Looking for 14166289921 in default (domain 66.28.190.219) list_route: hop: <--- Transmitting (NAT) to 72.187.243.97:45990 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 72.187.243.97:45990;branch=z9hG4bK137F96CF17EE4DC2AF466F1A35F17BEB;received=72.187.243.97 From: 7864335989 ;tag=168603557 To: Call-ID: E0C19ACC-FCCB-4193-8907-6FA09F3A3870@192.168.1.111 CSeq: 12134 INVITE User-Agent: Cisco 3845 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------>  -- Executing [14166289921@default:1] NoOp("SIP/7864335989-07377ab0", "SIP") in new stack  -- Executing [14166289921@default:2] GotoIf("SIP/7864335989-07377ab0", "0?default|14166289921|300") in new stack  -- Executing [14166289921@default:3] Set("SIP/7864335989-07377ab0", "SIPIP=72.187.243.97") in new stack  -- Executing [14166289921@default:4] Verbose("SIP/7864335989-07377ab0", "From= 72.187.243.97 Number= 14166289921") in new stack From= 72.187.243.97 Number= 14166289921  -- Executing [14166289921@default:5] GotoIf("SIP/7864335989-07377ab0", "1?default|14166289921|entrada") in new stack  -- Goto (default,14166289921,7)  -- Executing [14166289921@default:7] Set("SIP/7864335989-07377ab0", "CALLERID(num)=7864335989") in new stack  -- Executing [14166289921@default:8] NoOp("SIP/7864335989-07377ab0", "Callerid: 7864335989") in new stack  -- Executing [14166289921@default:9] Dial("SIP/7864335989-07377ab0", "SIP/14166289921@67.110.179.253") in new stack Audio is at 66.28.190.219 port 8058 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 67.110.179.253:5060: INVITE sip:14166289921@67.110.179.253 SIP/2.0 Via: SIP/2.0/UDP 66.28.190.219:5060;branch=z9hG4bK15ffeda5 From: "7864335989" ;tag=as2e79df44 To: Contact: Call-ID: 2aa75e1f3e31cbce29e420f44397c778@minixel CSeq: 102 INVITE User-Agent: Cisco 3845 Max-Forwards: 70 Date: Thu, 01 May 2008 22:24:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 262 v=0 o=root 3973 3973 IN IP4 66.28.190.219 s=session c=IN IP4 66.28.190.219 t=0 0 m=audio 8058 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv ---  -- Called 14166289921@67.110.179.253 <--- SIP read from 67.110.179.253:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 66.28.190.219:5060;branch=z9hG4bK15ffeda5 From: "7864335989" ;tag=as2e79df44 To: Date: Thu, 01 May 2008 22:24:13 GMT Call-ID: 2aa75e1f3e31cbce29e420f44397c778@minixel Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow-Events: telephone-event Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 67.110.179.253:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 66.28.190.219:5060;branch=z9hG4bK15ffeda5 From: "7864335989" ;tag=as2e79df44 To: ;tag=420E83C-1008 Date: Thu, 01 May 2008 22:24:13 GMT Call-ID: 2aa75e1f3e31cbce29e420f44397c778@minixel Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Remote-Party-ID: ;party=called;screen=no;privacy=off Contact: Supported: replaces Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 303 v=0 o=CiscoSystemsSIP-GW-UserAgent 3030 6525 IN IP4 67.110.179.253 s=SIP Call c=IN IP4 66.234.181.143 t=0 0 m=audio 55408 RTP/AVP 18 101 c=IN IP4 66.234.181.143 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=silenceSupp:off - - - - <-------------> --- (16 headers 13 lines) --- Found RTP audio format 18 Found RTP audio format 101 [May 1 18:24:18] DEBUG[3997]: chan_sip.c:5208 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 66.234.181.143:55408 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 66.234.181.143:55408 list_route: hop: [May 1 18:24:18] DEBUG[3997]: chan_sip.c:5951 reqprep: Strict routing enforced for session 2aa75e1f3e31cbce29e420f44397c778@minixel set_destination: Parsing for address/port to send to set_destination: set destination to 67.110.179.253, port 5060 Transmitting (NAT) to 67.110.179.253:5060: ACK sip:14166289921@67.110.179.253:5060 SIP/2.0 Via: SIP/2.0/UDP 66.28.190.219:5060;branch=z9hG4bK7b1f72cc From: "7864335989" ;tag=as2e79df44 To: ;tag=420E83C-1008 Contact: Call-ID: 2aa75e1f3e31cbce29e420f44397c778@minixel CSeq: 102 ACK User-Agent: Cisco 3845 Max-Forwards: 70 Content-Length: 0 ---  -- SIP/67.110.179.253-b0041e30 answered SIP/7864335989-07377ab0 Audio is at 66.28.190.219 port 21810 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 72.187.243.97:45990 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 72.187.243.97:45990;branch=z9hG4bK137F96CF17EE4DC2AF466F1A35F17BEB;received=72.187.243.97 From: 7864335989 ;tag=168603557 To: ;tag=as3d103c2e Call-ID: E0C19ACC-FCCB-4193-8907-6FA09F3A3870@192.168.1.111 CSeq: 12134 INVITE User-Agent: Cisco 3845 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 3973 3973 IN IP4 66.28.190.219 s=session c=IN IP4 66.28.190.219 t=0 0 m=audio 21810 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------>  -- Packet2Packet bridging SIP/7864335989-07377ab0 and SIP/67.110.179.253-b0041e30 <--- SIP read from 72.187.243.97:45990 ---> ACK sip:14166289921@66.28.190.219 SIP/2.0 Via: SIP/2.0/UDP 72.187.243.97:45990;branch=z9hG4bK8ED80175255D4E2CBA1805A6FD2EF46E From: 7864335989 ;tag=168603557 To: ;tag=as3d103c2e Contact: Call-ID: E0C19ACC-FCCB-4193-8907-6FA09F3A3870@192.168.1.111 CSeq: 12134 ACK Max-Forwards: 70 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from 72.187.243.97:45990 ---> BYE sip:14166289921@66.28.190.219 SIP/2.0 Via: SIP/2.0/UDP 72.187.243.97:45990;branch=z9hG4bK057272E32F4448B4ABE4107952B96094 From: 7864335989 ;tag=168603557 To: ;tag=as3d103c2e Contact: Call-ID: E0C19ACC-FCCB-4193-8907-6FA09F3A3870@192.168.1.111 CSeq: 12135 BYE Max-Forwards: 70 User-Agent: X-PRO release 1105x Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 72.187.243.97 : 45990 (NAT) <--- Transmitting (NAT) to 72.187.243.97:45990 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 72.187.243.97:45990;branch=z9hG4bK057272E32F4448B4ABE4107952B96094;received=72.187.243.97 From: 7864335989 ;tag=168603557 To: ;tag=as3d103c2e Call-ID: E0C19ACC-FCCB-4193-8907-6FA09F3A3870@192.168.1.111 CSeq: 12135 BYE User-Agent: Cisco 3845 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Scheduling destruction of SIP dialog '2aa75e1f3e31cbce29e420f44397c778@minixel' in 32000 ms (Method: INVITE) [May 1 18:24:26] DEBUG[4033]: chan_sip.c:5951 reqprep: Strict routing enforced for session 2aa75e1f3e31cbce29e420f44397c778@minixel set_destination: Parsing for address/port to send to set_destination: set destination to 67.110.179.253, port 5060 Reliably Transmitting (NAT) to 67.110.179.253:5060: BYE sip:14166289921@67.110.179.253:5060 SIP/2.0 Via: SIP/2.0/UDP 66.28.190.219:5060;branch=z9hG4bK5110de71 From: "7864335989" ;tag=as2e79df44 To: ;tag=420E83C-1008 Call-ID: 2aa75e1f3e31cbce29e420f44397c778@minixel CSeq: 103 BYE User-Agent: Cisco 3845 Max-Forwards: 70 Content-Length: 0 ---  == Spawn extension (default, 14166289921, 9) exited non-zero on 'SIP/7864335989-07377ab0' <--- SIP read from 67.110.179.253:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 66.28.190.219:5060;branch=z9hG4bK5110de71 From: "7864335989" ;tag=as2e79df44 To: ;tag=420E83C-1008 Date: Thu, 01 May 2008 22:24:26 GMT Call-ID: 2aa75e1f3e31cbce29e420f44397c778@minixel Server: Cisco-SIPGateway/IOS-12.x CSeq: 103 BYE Reason: Q.850;cause=16 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '2aa75e1f3e31cbce29e420f44397c778@minixel' Method: INVITE Really destroying SIP dialog 'E0C19ACC-FCCB-4193-8907-6FA09F3A3870@192.168.1.111' Method: BYE *CLI> stop now Beginning asterisk shutdown.... Executing last minute cleanups Asterisk cleanly ending (0).