Asterisk 1.6.1-rc1, 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 [Feb 2 18:32:31] NOTICE[29240]: loader.c:873 load_modules: 1 modules will be loaded.  == 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'  == Registered custom function 'TOUPPER'  == Registered custom function 'TOLOWER'  func_strings.so => (String handling dialplan functions)  == Parsing '/etc/asterisk/dnsmgr.conf':  == Found  == Parsing '/etc/asterisk/http.conf':  == Found  == Parsing '/etc/asterisk/cdr.conf':  == Found [Feb 2 18:32:31] NOTICE[29240]: cdr.c:1442 do_reload: CDR simple logging enabled.  == Parsing '/etc/asterisk/rtp.conf':  == Found  == RTP Allocating from port range 10000 -> 20000  == Parsing '/etc/asterisk/udptl.conf':  == Found  == UDPTL allocating from port range 4000 -> 4999  Asterisk PBX Core Initializing  Registering builtin applications:  == Registered custom function 'EXCEPTION'  [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'  [Incomplete]  == Registered application 'Incomplete'  [NoOp]  == Registered application 'NoOp'  [Proceeding]  == Registered application 'Proceeding'  [Progress]  == Registered application 'Progress'  [RaiseException]  == Registered application 'RaiseException'  [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'  [MSet]  == Registered application 'MSet'  [SetAMAFlags]  == Registered application 'SetAMAFlags'  [Wait]  == Registered application 'Wait'  [WaitExten]  == Registered application 'WaitExten'  == Manager registered action ShowDialPlan  == Registered application 'Bridge'  == Parsing '/etc/asterisk/features.conf':  == Found  == Registered application 'ParkedCall'  == Registered application 'Park'  == Manager registered action ParkedCalls  == Manager registered action Park  == Manager registered action Bridge  == 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 [Feb 2 18:32:31] NOTICE[29240]: loader.c:873 load_modules: 155 modules will be loaded.  == Registered custom function 'PP_EACH_USER'  == Registered custom function 'PP_EACH_EXTENSION'  == Parsing '/etc/asterisk/sip.conf':  == Found  == Parsing '/etc/asterisk/users.conf':  == Found  == Parsing '/etc/asterisk/phoneprov.conf':  == Found  res_phoneprov.so => (HTTP Phone Provisioning)  == Parsing '/etc/asterisk/res_ldap.conf':  == Found [Feb 2 18:32:31] WARNING[29240]: res_config_ldap.c:1401 parse_config: No directory user found, anonymous binding as default. [Feb 2 18:32:31] ERROR[29240]: res_config_ldap.c:1426 parse_config: No directory URL or host found. [Feb 2 18:32:31] NOTICE[29240]: res_config_ldap.c:1319 load_module: Cannot load LDAP RealTime driver.  res_config_ldap.so => (LDAP realtime interface)  == Parsing '/etc/asterisk/indications.conf':  == Found  == Registered application 'PlayTones'  == Registered application 'StopPlayTones'  res_indications.so => (Region-specific tones)  res_ael_share.so => (share-able code for AEL)  == 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)  res_speech.so => (Generic Speech Recognition API)  == Registered custom function 'SMDI_MSG_RETRIEVE'  == Registered custom function 'SMDI_MSG'  == Parsing '/etc/asterisk/smdi.conf':  == Found [Feb 2 18:32:31] NOTICE[29240]: res_smdi.c:1330 load_module: No SMDI interfaces are available to listen on, not starting SMDI listener.  == 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  == AGI Command 'speech create' registered  == AGI Command 'speech set' registered  == AGI Command 'speech destroy' registered  == AGI Command 'speech load grammar' registered  == AGI Command 'speech unload grammar' registered  == AGI Command 'speech activate grammar' registered  == AGI Command 'speech deactivate grammar' registered  == AGI Command 'speech recognize' registered  == Registered application 'DeadAGI'  == Registered application 'EAGI'  == Manager registered action AGI  == Registered application 'AGI'  res_agi.so => (Asterisk Gateway Interface (AGI))  == Parsing '/etc/asterisk/musiconhold.conf':  == Found  == Registered application 'MusicOnHold'  == Registered application 'WaitMusicOnHold'  == Registered application 'SetMusicOnHold'  == Registered application 'StartMusicOnHold'  == Registered application 'StopMusicOnHold'  res_musiconhold.so => (Music On Hold Resource)  == Registered application 'ChannelRedirect'  app_channelredirect.so => (Redirects a given channel to a dialplan target)  == Registered custom function 'AST_CONFIG'  func_config.so => (Asterisk configuration file variable access)  == Registered application 'TrySystem'  == Registered application 'System'  app_system.so => (Generic System() application)  == Registered application 'Log'  == Registered application 'Verbose'  app_verbose.so => (Send verbose output)  == Parsing '/etc/asterisk/codecs.conf':  == Found  == Registered translator 'lpc10tolin' from format lpc10 to slin, cost 1000  == Registered translator 'lintolpc10' from format slin to lpc10, cost 1000  codec_lpc10.so => (LPC10 2.4kbps Coder/Decoder)  == Registered application 'Transfer'  app_transfer.so => (Transfers a caller to another extension)  == Registered application 'Record'  app_record.so => (Trivial Record Application)  == Registered application 'Zapateller'  app_zapateller.so => (Block Telemarketers with Special Information Tone)  == Registered application 'ADSIProg'  app_adsiprog.so => (Asterisk ADSI Programming Application)  == Registered custom function 'SYSINFO'  func_sysinfo.so => (System information related functions)  == Parsing '/etc/asterisk/mgcp.conf':  == Found  == MGCP Listening on 0.0.0.0:2727  == Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP))  chan_mgcp.so => (Media Gateway Control Protocol (MGCP))  pbx_spool.so => (Outgoing Spool Support)  == Registered custom function 'IAXPEER'  == Registered custom function 'IAXVAR'  == Registered application 'IAX2Provision'  == Manager registered action IAXpeers  == Manager registered action IAXpeerlist  == Manager registered action IAXnetstats  == Parsing '/etc/asterisk/iax.conf':  == Found  == Parsing '/etc/asterisk/users.conf':  == Found  == 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  chan_iax2.so => (Inter Asterisk eXchange (Ver 2))  == Registered application 'Exec'  == Registered application 'TryExec'  == Registered application 'ExecIf'  app_exec.so => (Executes dialplan applications)  == Registered application 'MacroExit'  == Registered application 'MacroIf'  == Registered application 'MacroExclusive'  == Registered application 'Macro'  app_macro.so => (Extension Macros)  res_curl.so => (cURL Resource Module)  == Registered custom function 'CURL'  func_curl.so => (Load external URL)  == Registered custom function 'MATH'  func_math.so => (Mathematical dialplan function)  == Registered custom function 'DIALGROUP'  func_dialgroup.so => (Dialgroup dialplan function)  == Registered application 'Read'  app_read.so => (Read Variable Application)  == Registered application 'SMS'  app_sms.so => (SMS/PSTN handler)  == 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)  == Parsing '/etc/asterisk/extensions.conf':  == Found  == Setting global variable 'CONSOLE' to 'Console/dsp'  == Setting global variable 'IAXINFO' to 'guest'  == Setting global variable 'TRUNK' to 'DAHDI/G2'  == Setting global variable 'TRUNKMSD' to '1'  == Parsing '/etc/asterisk/users.conf':  == Found  pbx_config.so => (Text Extension Configuration)  == Manager registered action PlayDTMF  == Registered application 'SendDTMF'  app_senddtmf.so => (Send DTMF digits Application)  == Parsing '/etc/asterisk/alarmreceiver.conf':  == Found  == Registered application 'AlarmReceiver'  app_alarmreceiver.so => (Alarm Receiver for Asterisk)  == Parsing '/etc/asterisk/cdr_manager.conf':  == Found  cdr_manager.so => (Asterisk Manager Interface CDR Backend)  == 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 custom function 'SHA1'  func_sha1.so => (SHA-1 computation dialplan function)  == Parsing '/etc/asterisk/followme.conf':  == Found  == Registered application 'FollowMe'  app_followme.so => (Find-Me/Follow-Me Application)  == Registered application 'Directory'  app_directory.so => (Extension Directory)  == Registered application 'PrivacyManager'  app_privacy.so => (Require phone number to be entered, if no CallerID sent)  == Parsing '/etc/asterisk/adsi.conf':  == Found  res_adsi.so => (ADSI Resource)  == Registered custom function 'LOCK'  == Registered custom function 'TRYLOCK'  == Registered custom function 'UNLOCK'  func_lock.so => (Dialplan mutexes) [Feb 2 18:32:31] WARNING[29240]: loader.c:656 load_resource: Module 'res_curl.so' already exists.  == Registered application 'ReadExten'  == Registered custom function 'VALID_EXTEN'  app_readexten.so => (Read and evaluate extension validity)  == Registered file format ogg_vorbis, extension(s) ogg  format_ogg_vorbis.so => (OGG/Vorbis audio) [Feb 2 18:32:31] NOTICE[29240]: pbx_ael.c:122 pbx_load_module: Starting AEL load process. [Feb 2 18:32:31] NOTICE[29240]: pbx_ael.c:135 pbx_load_module: AEL load process: parsed config file name '/etc/asterisk/extensions.ael'. [Feb 2 18:32:31] NOTICE[29240]: pbx_ael.c:138 pbx_load_module: AEL load process: checked config file name '/etc/asterisk/extensions.ael'.  == Setting global variable 'CONSOLE' to '"Console/dsp"'  == Setting global variable 'IAXINFO' to 'guest'  == Setting global variable 'TRUNK' to '"DAHDI/G2"'  == Setting global variable 'TRUNKMSD' to '1' [Feb 2 18:32:31] NOTICE[29240]: pbx_ael.c:141 pbx_load_module: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'. [Feb 2 18:32:31] NOTICE[29240]: pbx_ael.c:146 pbx_load_module: AEL load process: merged config file name '/etc/asterisk/extensions.ael'. [Feb 2 18:32:31] NOTICE[29240]: pbx_ael.c:149 pbx_load_module: AEL load process: verified config file name '/etc/asterisk/extensions.ael'.  pbx_ael.so => (Asterisk Extension Language Compiler)  == Registered custom function 'TIMEOUT'  func_timeout.so => (Channel timeout 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 custom function 'RAND'  func_rand.so => (Random number dialplan function)  == Registered custom function 'IFMODULE'  func_module.so => (Checks if Asterisk module is loaded in memory)  == Registered application 'WaitUntil'  app_waituntil.so => (Wait until specified time)  == Parsing '/etc/asterisk/unistim.conf':  == Found  == UNISTIM Listening on 0.0.0.0:5000  == Registered channel type 'USTM' (UNISTIM Channel Driver)  chan_unistim.so => (UNISTIM Protocol (USTM))  == Registered custom function 'REALTIME'  == Registered custom function 'REALTIME_STORE'  == Registered custom function 'REALTIME_DESTROY'  func_realtime.so => (Read/Write/Store/Destroy values from a RealTime repository)  == 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 'MP3Player'  app_mp3.so => (Silly MP3 Application)  == Registered custom function 'CHANNEL'  == Registered custom function 'CHANNELS'  func_channel.so => (Channel information dialplan functions)  == Registered custom function 'BASE64_ENCODE'  == Registered custom function 'BASE64_DECODE'  func_base64.so => (base64 encode/decode dialplan functions)  == Registered custom function 'VOLUME'  func_volume.so => (Technology independent volume control)  == Parsing '/etc/asterisk/amd.conf':  == Found  == 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 application 'MinivmRecord'  == Registered application 'MinivmGreet'  == Registered application 'MinivmNotify'  == Registered application 'MinivmDelete'  == Registered application 'MinivmAccMess'  == Registered custom function 'MINIVMACCOUNT'  == Registered custom function 'MINIVMCOUNTER'  == Parsing '/etc/asterisk/minivm.conf':  == Found  app_minivm.so => (Mini VoiceMail (A minimal Voicemail e-mail System))  == 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 custom function 'SHELL'  func_shell.so => (Returns the output of a shell command)  == Registered file format h263, extension(s) h263  format_h263.so => (Raw H.263 data)  == Registered application 'ICES'  app_ices.so => (Encode and Stream via icecast and ices)  == Registered file format g723sf, extension(s) g723|g723sf  format_g723.so => (G.723.1 Simple Timestamp File Format)  res_limit.so => (Resource limits)  res_timing_pthread.so => (pthread Timing Interface)  == Registered custom function 'CDR'  func_cdr.so => (Call Detail Record (CDR) dialplan function)  == Parsing '/etc/asterisk/festival.conf':  == Found  == Registered application 'Festival'  app_festival.so => (Simple Festival Interface)  == Parsing '/etc/asterisk/oss.conf':  == Found  == Registered channel type 'Console' (OSS Console Channel Driver)  chan_oss.so => (OSS Console Channel Driver)  == Registered file format wav, extension(s) wav  format_wav.so => (Microsoft WAV format (8000Hz Signed Linear))  res_crypto.so => (Cryptographic Digital Signatures)  == Parsing '/etc/asterisk/say.conf':  == Found  == Registered application 'Playback'  app_playback.so => (Sound File Playback Application)  == 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 'g722tolin' from format g722 to slin, cost 1000  == Registered translator 'lintog722' from format slin to g722, cost 1 [Feb 2 18:32:31] WARNING[29240]: translate.c:645 __ast_register_translator: plc_samples 160 format f  == Registered translator 'g722tolin16' from format g722 to slin16, cost 1000  == Registered translator 'lin16tog722' from format slin16 to g722, cost 1000  codec_g722.so => (ITU G.722-64kbps G722 Transcoder)  == Registered application 'SayUnixTime'  == Registered application 'DateTime'  app_sayunixtime.so => (Say time)  == Registered application 'Milliwatt'  app_milliwatt.so => (Digital Milliwatt (mu-law) Test Application)  == Registered custom function 'CUT'  == Registered custom function 'SORT'  func_cut.so => (Cut out information from a string)  == Registered application 'SendImage'  app_image.so => (Image Transmission Application)  == Registered application 'Echo'  app_echo.so => (Simple Echo Application)  == Registered application 'Authenticate'  app_authenticate.so => (Authentication Application)  == Registered custom function 'CALLERPRES'  == Registered custom function 'CALLERID'  func_callerid.so => (Caller ID related dialplan functions)  == Parsing '/etc/asterisk/queuerules.conf':  == Found  == 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 QueueSummary  == Manager registered action QueueAdd  == Manager registered action QueueRemove  == Manager registered action QueuePause  == Manager registered action QueueLog  == Manager registered action QueuePenalty  == Manager registered action QueueRule  == Registered custom function 'QUEUE_VARIABLES'  == Registered custom function 'QUEUE_MEMBER'  == Registered custom function 'QUEUE_MEMBER_COUNT'  == Registered custom function 'QUEUE_MEMBER_LIST'  == Registered custom function 'QUEUE_WAITING_COUNT'  == Registered custom function 'QUEUE_MEMBER_PENALTY'  app_queue.so => (True Call Queueing)  == Registered application 'While'  == Registered application 'EndWhile'  == Registered application 'ExitWhile'  == Registered application 'ContinueWhile'  app_while.so => (While Loops and Conditional Execution)  pbx_realtime.so => (Realtime Switch)  == Registered custom function 'MD5'  func_md5.so => (MD5 digest dialplan functions)  == 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)) [Feb 2 18:32:31] NOTICE[29240]: config.c:1952 ast_config_engine_register: Registered Config Engine curl  res_config_curl loaded.  res_config_curl.so => (Realtime Curl configuration)  == Registered application 'WaitForRing'  app_waitforring.so => (Waits until first ring after time)  == Registered file format sln16, extension(s) sln16  format_sln16.so => (Raw Signed Linear 16KHz Audio support (SLN16))  == Registered channel type 'Local' (Local Proxy Channel Driver)  chan_local.so => (Local Proxy Channel (Note: used internally by other modules))  == Registered application 'NBScat'  app_nbscat.so => (Silly NBS Stream Application)  == Registered file format h264, extension(s) h264  format_h264.so => (Raw H.264 data)  == Parsing '/etc/asterisk/cdr_sqlite3_custom.conf':  == Found  cdr_sqlite3_custom.so => (SQLite3 Custom CDR Module)  == Registered application 'SetCallerPres'  app_setcallerid.so => (Set CallerID Presentation Application)  == Registered application 'DBdel'  == Registered application 'DBdeltree'  app_db.so => (Database Access Functions)  == Registered application 'UserEvent'  app_userevent.so => (Custom User Event Application)  == Registered application 'WaitForSilence'  == Registered application 'WaitForNoise'  app_waitforsilence.so => (Wait For Silence)  == Registered custom function 'ENV'  == Registered custom function 'STAT'  == Registered custom function 'FILE'  func_env.so => (Environment/filesystem dialplan functions)  == Registered custom function 'DEVICE_STATE'  == Registered custom function 'HINT'  func_devstate.so => (Gets or sets a device state in the dialplan)  == Registered custom function 'VERSION'  func_version.so => (Get Asterisk Version/Build Info)  == Registered application 'SendURL'  app_url.so => (Send URL Applications)  == 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 1000  codec_gsm.so => (GSM Coder/Decoder)  == Registered application 'ForkCDR'  app_forkcdr.so => (Fork The CDR into 2 separate entities)  == Registered custom function 'VMCOUNT'  func_vmcount.so => (Indicator for whether a voice mailbox has messages in a given folder.)  == Registered file format g729, extension(s) g729  format_g729.so => (Raw G729 data)  == Registered application 'ParkAndAnnounce'  app_parkandannounce.so => (Call Parking and Announce Application)  == Registered file format iLBC, extension(s) ilbc  format_ilbc.so => (Raw iLBC data)  == Registered file format sln, extension(s) sln|raw  format_sln.so => (Raw Signed Linear Audio support (SLN))  == Registered application 'NoCDR'  app_cdr.so => (Tell Asterisk to not maintain a CDR for the current call)  res_clioriginate.so => (Call origination from the CLI)  == Registered custom function 'DIALPLAN_EXISTS'  func_dialplan.so => (Dialplan Context/Extension/Priority Checking Functions)  == Registered custom function 'ICONV'  func_iconv.so => (Charset conversions)  == Registered application 'ReadFile'  app_readfile.so => (Stores output of file into a variable)  == Parsing '/etc/asterisk/cdr_custom.conf':  == Found  cdr_custom.so => (Customizable Comma Separated Values CDR Backend)  == Registered application 'GetCPEID'  app_getcpeid.so => (Get ADSI CPE ID)  == Registered custom function 'BLACKLIST'  func_blacklist.so => (Look up Caller*ID name/number from blacklist database)  == Registered application 'ControlPlayback'  app_controlplayback.so => (Control Playback Application)  == Registered format 'jpg' (JPEG (Joint Picture Experts Group))  format_jpeg.so => (JPEG (Joint Picture Experts Group) Image Format)  == Registered custom function 'ENUMRESULT'  == Registered custom function 'ENUMQUERY'  == Registered custom function 'ENUMLOOKUP'  == Registered custom function 'TXTCIDNAME'  func_enum.so => (ENUM related dialplan functions) SIP channel loading...  == Parsing '/etc/asterisk/sip.conf':  == Found  == Parsing '/etc/asterisk/users.conf':  == Found  == SIP Listening on asterisk.com:5060  == Using SIP CoS mark 4 SSL certificate ok  == 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  == Manager registered action SIPqualifypeer  == Manager registered action SIPshowregistry  == Manager registered action SIPnotify  chan_sip.so => (Session Initiation Protocol (SIP))  == Registered application 'Pickup'  == Registered application 'PickupChan'  app_directed_pickup.so => (Directed Call Pickup Application)  == Registered application 'ChanSpy'  == Registered application 'ExtenSpy'  app_chanspy.so => (Listen to the audio of an active channel)  == Registered custom function 'AUDIOHOOK_INHERIT'  func_audiohookinherit.so => (Audiohook inheritance function)  == Registered application 'DumpChan'  app_dumpchan.so => (Dump Info About The Calling Channel)  == Registered custom function 'URIDECODE'  == Registered custom function 'URIENCODE'  func_uri.so => (URI encode/decode 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)  == 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)  == 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 1000  == Registered translator 'g726aal2tolin' from format g726aal2 to slin, cost 1000  == Registered translator 'lintog726aal2' from format slin to g726aal2, cost 1000  == 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)  res_realtime.so => (Realtime Data Lookup/Rewrite)  == Parsing '/etc/asterisk/dundi.conf':  == Found  == 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))  == Parsing '/etc/asterisk/phone.conf':  == Found  == Registered channel type 'Phone' (Standard Linux Telephony API Driver)  chan_phone.so => (Linux Telephony API Support)  == Registered application 'ChanIsAvail'  app_chanisavail.so => (Check channel availability)  == Registered application 'Morsecode'  app_morsecode.so => (Morse code)  == Registered application 'TestClient'  == Registered application 'TestServer'  app_test.so => (Interface Test Application)  res_convert.so => (File format conversion CLI command)  == 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 'ExternalIVR'  app_externalivr.so => (External IVR Interface Application)  == 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)  == Registered custom function 'EXTENSION_STATE'  func_extstate.so => (Gets an extension's state in the dialplan)  == Registered application 'MixMonitor'  == Registered application 'StopMixMonitor'  app_mixmonitor.so => (Mixed Audio Monitoring 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 application 'Dial'  == Registered application 'RetryDial'  app_dial.so => (Dialing Application)  == Registered custom function 'GLOBAL'  == Registered custom function 'SHARED'  func_global.so => (Variable dialplan functions)  == Registered application 'SoftHangup'  app_softhangup.so => (Hangs up the requested channel)  == Parsing '/etc/asterisk/cdr.conf':  == Found  cdr_csv.so => (Comma Separated Values CDR Backend)  == Registered application 'SendText'  app_sendtext.so => (Send Text Applications)  == 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 file format vox, extension(s) vox  format_vox.so => (Dialogic VOX (ADPCM) File Format)  == AGI Command 'gosub' registered  == 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 file format gsm, extension(s) gsm  format_gsm.so => (Raw GSM data)  == Registered application 'Dictate'  app_dictate.so => (Virtual Dictation Machine)  pbx_loopback.so => (Loopback Switch)  == Registered application 'DISA'  app_disa.so => (DISA (Direct Inward System Access) Application)  == Registered application 'BackgroundDetect'  app_talkdetect.so => (Playback with Talk Detection)  == 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 CreateConfig  == Manager registered action ListCategories  == Manager registered action Redirect  == Manager registered action Atxfer  == 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  == Manager registered action Reload  == Manager registered action CoreShowChannels  == Manager registered action ModuleLoad  == Manager registered action ModuleCheck  == Parsing '/etc/asterisk/manager.conf':  == Found  == Parsing '/etc/asterisk/users.conf':  == Found Asterisk Ready.  == Parsing '/etc/asterisk/cli.conf':  == Found ]1;Asterisk]2;Asterisk Console on 'sipmac6-fc8-vm1' (pid 29240)*CLI> sip set debug on SIP Debugging enabled *CLI> <--- SIP read from TLS://snomclient.com:2139 ---> REGISTER sip:asterisk.com:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS snomclient.com:2139;branch=z9hG4bK-azw0uuow2h5q;rport From: "7780" ;tag=ctder1wfv3 To: "7780" Call-ID: 25f386493336-hkzr8tvkarws@snomSoft-000413FFFFFF CSeq: 1 REGISTER Max-Forwards: 70 Contact: ;q=1.0;flow-id=1;+sip.instance="";audio;mobility="fixed";duplex="full";description="snomSoft";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snomSoft/5.3 Supported: gruu Allow-Events: dialog X-Real-IP: snomclient.com Expires: 3600 Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Sending to snomclient.com : 2139 (NAT)  -- Registered SIP '7780' at snomclient.com port 2139 <--- Transmitting (no NAT) to snomclient.com:2139 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS snomclient.com:2139;branch=z9hG4bK-azw0uuow2h5q;received=snomclient.com;rport=2139 From: "7780" ;tag=ctder1wfv3 To: "7780" ;tag=as4ea7b972 Call-ID: 25f386493336-hkzr8tvkarws@snomSoft-000413FFFFFF CSeq: 1 REGISTER Server: Asterisk PBX 1.6.1-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Expires: 3600 Contact: ;expires=3600 Date: Mon, 02 Feb 2009 13:02:35 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '25f386493336-hkzr8tvkarws@snomSoft-000413FFFFFF' in 32000 ms (Method: REGISTER) <--- SIP read from TLS://MediationServer.com:2381 ---> INVITE sip:7780@asterisk.com;user=phone SIP/2.0 FROM: ;epid=8631B29785;tag=5bfad72862 TO: CSEQ: 12 INVITE CALL-ID: e9b61cd7-68f0-4b0a-bc52-a9e9748a35a3 MAX-FORWARDS: 70 VIA: SIP/2.0/TLS MediationServer.com:2381;branch=z9hG4bKcfe434ce CONTACT: CONTENT-LENGTH: 302 SUPPORTED: 100rel USER-AGENT: RTCC/3.0.0.0 MediationServer CONTENT-TYPE: application/sdp; charset=utf-8 ALLOW: UPDATE ALLOW: Ack, Cancel, Bye,Invite v=0 o=- 0 0 IN IP4 MediationServer.com s=session c=IN IP4 MediationServer.com b=CT:1000 t=0 0 m=audio 63208 RTP/AVP 97 101 0 8 c=IN IP4 MediationServer.com a=rtcp:63209 a=label:Audio a=rtpmap:97 RED/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=ptime:20 <-------------> --- (14 headers 16 lines) ---  == Using SIP RTP CoS mark 5 Sending to MediationServer.com : 2381 (no NAT) Using INVITE request as basis request - e9b61cd7-68f0-4b0a-bc52-a9e9748a35a3 No matching peer for '+14083335556' from 'MediationServer.com:2381' Found RTP audio format 97 Found RTP audio format 101 Found RTP audio format 0 Found RTP audio format 8 Peer audio RTP is at port MediationServer.com:63208 Found audio description format telephone-event for ID 101 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x40c (ulaw|alaw|ilbc)/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 MediationServer.com:63208 Looking for 7780 in default (domain asterisk.com) list_route: hop: <--- Transmitting (no NAT) to MediationServer.com:2381 ---> SIP/2.0 100 Trying Via: SIP/2.0/TLS MediationServer.com:2381;branch=z9hG4bKcfe434ce;received=MediationServer.com From: ;epid=8631B29785;tag=5bfad72862 To: Call-ID: e9b61cd7-68f0-4b0a-bc52-a9e9748a35a3 CSeq: 12 INVITE Server: Asterisk PBX 1.6.1-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> Audio is at asterisk.com port 17228 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to MediationServer.com:2381 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS MediationServer.com:2381;branch=z9hG4bKcfe434ce;received=MediationServer.com From: ;epid=8631B29785;tag=5bfad72862 To: ;tag=as5e8b9fac Call-ID: e9b61cd7-68f0-4b0a-bc52-a9e9748a35a3 CSeq: 12 INVITE Server: Asterisk PBX 1.6.1-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 289 v=0 o=root 1878093823 1878093823 IN IP4 asterisk.com s=Asterisk PBX 1.6.1-rc1 c=IN IP4 asterisk.com t=0 0 m=audio 17228 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> <--- SIP read from TLS://MediationServer.com:2381 ---> ACK sip:7780@asterisk.com:5060;transport=TLS SIP/2.0 FROM: ;epid=8631B29785;tag=5bfad72862 TO: ;tag=as5e8b9fac CSEQ: 12 ACK CALL-ID: e9b61cd7-68f0-4b0a-bc52-a9e9748a35a3 MAX-FORWARDS: 70 VIA: SIP/2.0/TLS MediationServer.com:2381;branch=z9hG4bK15ef6972 CONTENT-LENGTH: 0 USER-AGENT: RTCC/3.0.0.0 MediationServer <-------------> --- (9 headers 0 lines) ---  == Using SIP RTP CoS mark 5 Audio is at asterisk.com port 19174 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to snomclient.com:2139: INVITE sip:7780@snomclient.com:2139;transport=tls;line=ojn9itpa SIP/2.0 Via: SIP/2.0/TLS asterisk.com:5060;branch=z9hG4bK59e8c9e7;rport Max-Forwards: 70 From: "+14083335556" ;tag=as206d2003 To: Contact: Call-ID: 10e39dad1e14ae6d15a39641601c397e@asterisk.com CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1-rc1 Date: Mon, 02 Feb 2009 13:02:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 312 v=0 o=root 2146894028 2146894028 IN IP4 asterisk.com s=Asterisk PBX 1.6.1-rc1 c=IN IP4 asterisk.com t=0 0 m=audio 19174 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- <--- SIP read from TLS://snomclient.com:2139 ---> SUBSCRIBE sip:7780@asterisk.com:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS snomclient.com:2139;branch=z9hG4bK-xac9kkxm5nve;rport From: ;tag=kymf35yifu To: Call-ID: 25f386493336-gg4xst1mac3j@snomSoft-000413FFFFFF CSeq: 1 SUBSCRIBE Max-Forwards: 70 Contact: ;flow-id=1 Event: message-summary Accept: application/simple-message-summary Expires: 3600 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Creating new subscription Sending to snomclient.com : 2139 (NAT) Found peer '7780' for '7780' from snomclient.com:2139 Looking for 7780 in default (domain asterisk.com) <--- Transmitting (no NAT) to snomclient.com:2139 ---> SIP/2.0 404 Not Found Via: SIP/2.0/TLS snomclient.com:2139;branch=z9hG4bK-xac9kkxm5nve;received=snomclient.com;rport=2139 From: ;tag=kymf35yifu To: ;tag=as767716cc Call-ID: 25f386493336-gg4xst1mac3j@snomSoft-000413FFFFFF CSeq: 1 SUBSCRIBE Server: Asterisk PBX 1.6.1-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 <------------> <--- SIP read from TLS://snomclient.com:2139 ---> SIP/2.0 180 Ringing Via: SIP/2.0/TLS asterisk.com:5060;branch=z9hG4bK59e8c9e7;rport=5061 From: "+14083335556" ;tag=as206d2003 To: ;tag=f85md5vcqc Call-ID: 10e39dad1e14ae6d15a39641601c397e@asterisk.com CSeq: 102 INVITE Contact: ;flow-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '25f386493336-gg4xst1mac3j@snomSoft-000413FFFFFF' Method: SUBSCRIBE <--- SIP read from TLS://snomclient.com:2139 ---> SIP/2.0 200 Ok Via: SIP/2.0/TLS asterisk.com:5060;branch=z9hG4bK59e8c9e7;rport=5061 From: "+14083335556" ;tag=as206d2003 To: ;tag=f85md5vcqc Call-ID: 10e39dad1e14ae6d15a39641601c397e@asterisk.com CSeq: 102 INVITE Contact: ;flow-id=1 User-Agent: snomSoft/5.3 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 319 v=0 o=root 20443 20444 IN IP4 snomclient.com s=call c=IN IP4 snomclient.com t=0 0 m=audio 56942 RTP/AVP 0 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:UnG864mhqIjuuObiMYqM0WVbm4xxo8Gm3NxyEZM9 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv FO" <-------------> --- (13 headers 14 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port snomclient.com:56942 Found audio description format pcmu for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port snomclient.com:56942 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to snomclient.com, port 2139 Transmitting (no NAT) to snomclient.com:2139: ACK sip:7780@snomclient.com:2139;transport=tls;line=ojn9itpa SIP/2.0 Via: SIP/2.0/TLS asterisk.com:5060;branch=z9hG4bK1d49188a;rport Max-Forwards: 70 From: "+14083335556" ;tag=as206d2003 To: ;tag=f85md5vcqc Contact: Call-ID: 10e39dad1e14ae6d15a39641601c397e@asterisk.com CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1-rc1 Content-Length: 0 --- <--- SIP read from TLS://MediationServer.com:2381 ---> BYE sip:7780@asterisk.com:5060;transport=TLS SIP/2.0 FROM: ;epid=8631B29785;tag=5bfad72862 TO: ;tag=as5e8b9fac CSEQ: 13 BYE CALL-ID: e9b61cd7-68f0-4b0a-bc52-a9e9748a35a3 MAX-FORWARDS: 70 VIA: SIP/2.0/TLS MediationServer.com:2381;branch=z9hG4bK8a3bbcc1 CONTENT-LENGTH: 0 USER-AGENT: RTCC/3.0.0.0 MediationServer <-------------> --- (9 headers 0 lines) --- Sending to MediationServer.com : 2381 (no NAT) <--- Transmitting (no NAT) to MediationServer.com:2381 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS MediationServer.com:2381;branch=z9hG4bK8a3bbcc1;received=MediationServer.com From: ;epid=8631B29785;tag=5bfad72862 To: ;tag=as5e8b9fac Call-ID: e9b61cd7-68f0-4b0a-bc52-a9e9748a35a3 CSeq: 13 BYE Server: Asterisk PBX 1.6.1-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> Scheduling destruction of SIP dialog '10e39dad1e14ae6d15a39641601c397e@asterisk.com' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to snomclient.com, port 2139 Reliably Transmitting (no NAT) to snomclient.com:2139: BYE sip:7780@snomclient.com:2139;transport=tls;line=ojn9itpa SIP/2.0 Via: SIP/2.0/TLS asterisk.com:5060;branch=z9hG4bK144bfe49;rport Max-Forwards: 70 From: "+14083335556" ;tag=as206d2003 To: ;tag=f85md5vcqc Call-ID: 10e39dad1e14ae6d15a39641601c397e@asterisk.com CSeq: 103 BYE User-Agent: Asterisk PBX 1.6.1-rc1 Content-Length: 0 ---  == Spawn extension (default, 7780, 2) exited non-zero on 'SIP/MediationServer.com-09c87430' <--- SIP read from TLS://snomclient.com:2139 ---> SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/TLS asterisk.com:5060;branch=z9hG4bK144bfe49;rport=5061 From: "+14083335556" ;tag=as206d2003 To: ;tag=f85md5vcqc Call-ID: 10e39dad1e14ae6d15a39641601c397e@asterisk.com CSeq: 103 BYE Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Content-Length: 0 <-------------> --- (10 headers 0 lines) ---