[root@linuxserver boris]# sh ast-debug.sh Parsing /etc/asterisk/asterisk.conf Seeding global EID '00:07:e9:a5:3a:74' from 'eth0' using 'siocgifhwaddr' Parsing /etc/asterisk/logger.conf == Parsing '/etc/asterisk/asterisk.conf': == Found == Manager registered action DataGet == Parsing '/etc/asterisk/codecs.conf': == Found Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': == Found [Feb 19 18:02:50] NOTICE[30811]: loader.c:1118 load_modules: 1 modules will be loaded. == Parsing '/etc/asterisk/res_odbc.conf': == Found [Feb 19 18:02:50] NOTICE[30811]: res_odbc.c:1472 odbc_obj_connect: Connecting asterisk_mysql [Feb 19 18:02:51] NOTICE[30811]: res_odbc.c:1502 odbc_obj_connect: res_odbc: Connected to asterisk_mysql [asterisk] [Feb 19 18:02:51] NOTICE[30811]: res_odbc.c:900 load_odbc_config: Registered ODBC class 'asterisk_mysql' dsn->[asterisk] == Registered application 'ODBC_Commit' == Registered application 'ODBC_Rollback' == Registered custom function 'ODBC' [Feb 19 18:02:51] NOTICE[30811]: res_odbc.c:1830 load_module: res_odbc loaded. res_odbc.so => (ODBC resource) == 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 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 == Manager registered action AOCMessage == Parsing '/etc/asterisk/manager.conf': == Found == Parsing '/etc/asterisk/manager.d/README.conf': == Found == Parsing '/etc/asterisk/manager.d/op-panel.conf': == Found == Parsing '/etc/asterisk/users.conf': == Found == Parsing '/etc/asterisk/cdr.conf': == Found [Feb 19 18:02:51] NOTICE[30811]: cdr.c:1567 do_reload: CDR simple logging enabled. == Parsing '/etc/asterisk/cel.conf': == Found -- CEL logging disabled. == Parsing '/etc/asterisk/dsp.conf': == Found == Parsing '/etc/asterisk/udptl.conf': == Found == UDPTL allocating from port range 4000 -> 4999 Asterisk PBX Core Initializing Registering builtin applications: == Registered custom function 'EXCEPTION' == Registered custom function 'TESTTIME' [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 == 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 'ph' -- 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-old' -- Registered indication country 'tw' -- Registered indication country 've' -- Registered indication country 'za' -- Setting default indication country to 'ru' == Registered application 'Bridge' -- Registered extension context 'parkedcalls'; registrar: features -- Added extension '700' priority 1 to parkedcalls == Parsing '/etc/asterisk/features.conf': == Found -- Added extension '700' priority 1 to parkedcalls == 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 == Registered application 'CallCompletionRequest' == Registered application 'CallCompletionCancel' == Parsing '/etc/asterisk/ccss.conf': == Found Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': == Found [Feb 19 18:02:51] NOTICE[30811]: loader.c:1118 load_modules: 176 modules will be loaded. == 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_crypto.so => (Cryptographic Digital Signatures) == Parsing '/etc/asterisk/res_fax.conf': == Found == Registered application 'SendFAX' == Registered application 'ReceiveFAX' == Registered custom function 'FAXOPT' res_fax.so => (Generic FAX Applications) res_adsi => (ADSI Resource) res_speech.so => (Generic Speech Recognition API) == AGI Command 'answer' registered == AGI Command 'asyncagi break' 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/calendar.conf': == Found == Registered custom function 'CALENDAR_BUSY' == Registered custom function 'CALENDAR_EVENT' == Registered custom function 'CALENDAR_QUERY' == Registered custom function 'CALENDAR_QUERY_RESULT' == Registered custom function 'CALENDAR_WRITE' res_calendar.so => (Asterisk Calendar integration) res_ael_share.so => (share-able code for AEL) res_curl.so => (cURL Resource Module) == Registered custom function 'CURL' == Registered custom function 'CURLOPT' func_curl.so => (Load external URL) == 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 RTP engine 'asterisk' == Parsing '/etc/asterisk/rtp.conf': == Found == RTP Allocating from port range 10000 -> 20000 res_rtp_asterisk.so => (Asterisk RTP Stack) res_timing_pthread.so => (pthread Timing Interface) == Registered RTP engine 'multicast' res_rtp_multicast.so => (Multicast RTP Engine) == Registered application 'DAHDISendKeypadFacility' == Registered application 'DAHDISendCallreroutingFacility' == Parsing '/etc/asterisk/chan_dahdi.conf': == Found == Parsing '/etc/asterisk/users.conf': == Found -- Automatically generated pseudo channel == Registered channel type 'DAHDI' (DAHDI Telephony Driver w/PRI & SS7 & MFC/R2) == Registered application 'DAHDIAcceptR2Call' == Manager registered action DAHDITransfer == Manager registered action DAHDIHangup == Manager registered action DAHDIDialOffhook == Manager registered action DAHDIDNDon == Manager registered action DAHDIDNDoff == Manager registered action DAHDIShowChannels == Manager registered action DAHDIRestart chan_dahdi.so => (DAHDI Telephony Driver w/PRI & SS7 & MFC/R2) == Registered channel type 'Bridge' (Bridge Interaction Channel) chan_bridge.so => (Bridge Interaction Channel) SIP channel loading... == Parsing '/etc/asterisk/sip.conf': == Found == Parsing '/etc/asterisk/users.conf': == Found == SIP Listening on 192.168.70.51:5060 == Using SIP TOS bits 104 == Using SIP CoS mark 3 == Registered channel type 'SIP' (Session Initiation Protocol (SIP)) == Registered RTP glue 'SIP' == Registered application 'SIPDtmfMode' == Registered application 'SIPAddHeader' == Registered application 'SIPRemoveHeader' == 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 channel type 'MulticastRTP' (Multicast RTP Paging Channel Driver) chan_multicast_rtp.so => (Multicast RTP Paging Channel) == Registered channel type 'Local' (Local Proxy Channel Driver) == Manager registered action LocalOptimizeAway chan_local.so => (Local Proxy Channel (Note: used internally by other modules)) == Registered file format sln16, extension(s) sln16 format_sln16.so => (Raw Signed Linear 16KHz Audio support (SLN16)) == Registered file format sln, extension(s) sln|raw format_sln.so => (Raw Signed Linear Audio support (SLN)) == Registered file format wav49, extension(s) WAV|wav49 format_wav_gsm.so => (Microsoft WAV format (Proprietary GSM)) == Registered file format pcm, extension(s) pcm|ulaw|ul|mu|ulw == Registered file format alaw, extension(s) alaw|al|alw == 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 file format g729, extension(s) g729 format_g729.so => (Raw G.729 data) == Registered file format vox, extension(s) vox format_vox.so => (Dialogic VOX (ADPCM) File Format) == Registered file format g719, extension(s) g719 format_g719.so => (ITU G.719) == Registered file format wav, extension(s) wav == Registered file format wav16, extension(s) wav16 format_wav.so => (Microsoft WAV/WAV16 format (8kHz/16kHz Signed Linear)) == Registered file format ogg_vorbis, extension(s) ogg format_ogg_vorbis.so => (OGG/Vorbis audio) == Registered file format h263, extension(s) h263 format_h263.so => (Raw H.263 data) == Registered file format gsm, extension(s) gsm format_gsm.so => (Raw GSM data) == Registered file format g723sf, extension(s) g723|g723sf format_g723.so => (G.723.1 Simple Timestamp File Format) == 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 file format h264, extension(s) h264 format_h264.so => (Raw H.264 data) == Registered file format iLBC, extension(s) ilbc format_ilbc.so => (Raw iLBC data) == Registered file format siren7, extension(s) siren7 format_siren7.so => (ITU G.722.1 (Siren7, licensed from Polycom)) == Registered file format siren14, extension(s) siren14 format_siren14.so => (ITU G.722.1 Annex C (Siren14, licensed from Polycom)) == Registered format 'jpg' (JPEG (Joint Picture Experts Group)) format_jpeg.so => (jpeg (joint picture experts group) image format) == Registered application 'ConfBridge' app_confbridge.so => (Conference Bridge Application) == Registered custom function 'DEVICE_STATE' == Registered custom function 'HINT' func_devstate.so => (Gets or sets a device state in the dialplan) == Parsing '/etc/asterisk/meetme.conf': == Found == Manager registered action MeetmeMute == Manager registered action MeetmeUnmute == Manager registered action MeetmeList == Registered application 'MeetMeChannelAdmin' == Registered application 'MeetMeAdmin' == Registered application 'MeetMeCount' == Registered application 'MeetMe' == Registered application 'SLAStation' == Registered application 'SLATrunk' == Registered custom function 'MEETME_INFO' app_meetme.so => (MeetMe conference bridge) == Parsing '/etc/asterisk/cdr.conf': == Found cdr_csv.so => (Comma Separated Values CDR Backend) == Parsing '/etc/asterisk/cdr_manager.conf': == Found cdr_manager.so => (Asterisk Manager Interface CDR Backend) == Parsing '/etc/asterisk/cel_odbc.conf': == Found cel_odbc.so => (ODBC CEL backend) == Parsing '/etc/asterisk/cdr_odbc.conf': == Found -- cdr_odbc: dsn is asterisk_mysql -- cdr_odbc: table is cdr cdr_odbc.so => (ODBC CDR Backend) == Parsing '/etc/asterisk/cel.conf': == Found cel_manager.so => (Asterisk Manager Interface CEL Backend) == Parsing '/etc/asterisk/cel_custom.conf': == Found cel_custom.so => (Customizable Comma Separated Values CEL Backend) res_realtime.so => (Realtime Data Lookup/Rewrite) == Registered custom function 'CALLERPRES' == Registered custom function 'CALLERID' == Registered custom function 'CONNECTEDLINE' == Registered custom function 'REDIRECTING' func_callerid.so => (Party ID related dialplan functions (Caller-ID, Connected-line, Redirecting)) pbx_spool.so => (Outgoing Spool Support) == Registered bridge technology multiplexed_bridge bridge_multiplexed.so => (Multiplexed two channel bridging module) == Registered custom function 'IFMODULE' func_module.so => (Checks if Asterisk module is loaded in memory) == Registered application 'ForkCDR' app_forkcdr.so => (Fork The CDR into 2 separate entities) == 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 application 'VMSayName' == Registered custom function 'MAILBOX_EXISTS' == Manager registered action VoicemailUsersList app_voicemail.so => (Comedian Mail (Voicemail System)) == Registered custom function 'MD5' func_md5.so => (MD5 digest dialplan functions) == Registered custom function 'AUDIOHOOK_INHERIT' func_audiohookinherit.so => (Audiohook inheritance function) == Registered custom function 'BASE64_ENCODE' == Registered custom function 'BASE64_DECODE' func_base64.so => (base64 encode/decode dialplan functions) == Registered application 'SendImage' app_image.so => (Image Transmission Application) == Registered application 'MacroExit' == Registered application 'MacroIf' == Registered application 'MacroExclusive' == Registered application 'Macro' app_macro.so => (Extension Macros) res_convert.so => (File format conversion CLI command) == Registered translator 'adpcmtolin' from format adpcm to slin, cost 1000 == Registered translator 'lintoadpcm' from format slin to adpcm, cost 1000 codec_adpcm.so => (Adaptive Differential PCM Coder/Decoder) == Registered custom function 'VMCOUNT' func_vmcount.so => (Indicator for whether a voice mailbox has messages in a given folder.) == Registered application 'GetCPEID' app_getcpeid.so => (Get ADSI CPE ID) == Registered application 'DumpChan' app_dumpchan.so => (Dump Info About The Calling Channel) == Registered application 'DAHDIRAS' app_dahdiras.so => (DAHDI ISDN Remote Access Server) == Registered application 'Read' app_read.so => (Read Variable Application) == Registered application 'MP3Player' app_mp3.so => (Silly MP3 Application) == Registered translator 'g722tolin' from format g722 to slin, cost 3000 == Registered translator 'lintog722' from format slin to g722, cost 2000 == Registered translator 'g722tolin16' from format g722 to slin16, cost 3000 == Registered translator 'lin16tog722' from format slin16 to g722, cost 3999 codec_g722.so => (ITU G.722-64kbps G722 Transcoder) -- Security Logging Enabled res_security_log.so => (Security Event Logging) == Registered custom function 'MATH' == Registered custom function 'INC' == Registered custom function 'DEC' func_math.so => (Mathematical dialplan function) == Registered application 'Zapateller' app_zapateller.so => (Block Telemarketers with Special Information Tone) == Manager registered action PlayDTMF == Registered application 'SendDTMF' app_senddtmf.so => (Send DTMF digits Application) == Registered application 'ReadExten' == Registered custom function 'VALID_EXTEN' app_readexten.so => (Read and evaluate extension validity) == Registered application 'DBdel' == Registered application 'DBdeltree' app_db.so => (Database Access Functions) == Registered custom function 'PITCH_SHIFT' func_pitchshift.so => (Audio Effects Dialplan Functions) == Parsing '/etc/asterisk/followme.conf': == Found == Registered application 'FollowMe' app_followme.so => (Find-Me/Follow-Me Application) == Registered custom function 'VERSION' func_version.so => (Get Asterisk Version/Build Info) == Registered custom function 'ENV' == Registered custom function 'STAT' == Registered custom function 'FILE' == Registered custom function 'FILE_COUNT_LINE' == Registered custom function 'FILE_FORMAT' func_env.so => (Environment/filesystem dialplan functions) == Registered custom function 'MUTEAUDIO' == Manager registered action MuteAudio res_mutestream.so => (Mute audio stream resources) == Registered application 'MixMonitor' == Registered application 'StopMixMonitor' == Manager registered action MixMonitorMute app_mixmonitor.so => (Mixed Audio Monitoring Application) == Registered custom function 'CDR' func_cdr.so => (Call Detail Record (CDR) dialplan function) == Registered custom function 'SPRINTF' func_sprintf.so => (SPRINTF dialplan function) == Registered translator 'lpc10tolin' from format lpc10 to slin, cost 4999 == Registered translator 'lintolpc10' from format slin to lpc10, cost 5999 codec_lpc10.so => (LPC10 2.4kbps Coder/Decoder) == Registered bridge technology simple_bridge bridge_simple.so => (Simple two channel bridging module) == Parsing '/etc/asterisk/cli_aliases.conf': == Found == Aliased CLI command 'hangup request' to 'channel request hangup' == Aliased CLI command 'originate' to 'channel originate' == Aliased CLI command 'help' to 'core show help' == Aliased CLI command 'pri intense debug span' to 'pri set debug 2 span' == Aliased CLI command 'reload' to 'module reload' == Aliased CLI command 'shutdown' to 'core stop gracefully' == Aliased CLI command 'shutdown now' to 'core stop now' res_clialiases.so => (CLI Aliases) == Registered application 'Transfer' app_transfer.so => (Transfers a caller to another extension) == Registered custom function 'REALTIME' == Registered custom function 'REALTIME_STORE' == Registered custom function 'REALTIME_DESTROY' == Registered custom function 'REALTIME_FIELD' == Registered custom function 'REALTIME_HASH' func_realtime.so => (Read/Write/Store/Destroy values from a RealTime repository) == Registered custom function 'AES_DECRYPT' == Registered custom function 'AES_ENCRYPT' func_aes.so => (AES dialplan functions) == Registered custom function 'CALLCOMPLETION' func_callcompletion.so => (Call Control Configuration Function) == Registered custom function 'EXTENSION_STATE' func_extstate.so => (Gets an extension's state in the dialplan) == Registered application 'Pickup' == Registered application 'PickupChan' app_directed_pickup.so => (Directed Call Pickup Application) -- Registered extension context 'app_dial_gosub_virtual_context'; registrar: app_dial -- Added extension 's' priority 1 to app_dial_gosub_virtual_context == Registered application 'Dial' == Registered application 'RetryDial' app_dial.so => (Dialing Application) == Registered application 'PlayTones' == Registered application 'StopPlayTones' app_playtones.so => (Playtones Application) == Registered translator 'alawtoulaw' from format alaw to ulaw, cost 1000 == Registered translator 'ulawtoalaw' from format ulaw to alaw, cost 1 codec_a_mu.so => (A-law and Mulaw direct Coder/Decoder) pbx_realtime.so => (Realtime Switch) pbx_loopback.so => (Loopback Switch) == Registered application 'Echo' app_echo.so => (Simple Echo Application) bridge_builtin_features.so => (Built in bridging features) == Registered application 'SayUnixTime' == Registered application 'DateTime' app_sayunixtime.so => (Say time) == Registered custom function 'DIALGROUP' func_dialgroup.so => (Dialgroup dialplan function) == Registered application 'ChanSpy' == Registered application 'ExtenSpy' == Registered application 'DAHDIScan' app_chanspy.so => (Listen to the audio of an active channel) == Registered application 'PrivacyManager' app_privacy.so => (Require phone number to be entered, if no CallerID sent) == Registered application 'TestClient' == Registered application 'TestServer' app_test.so => (Interface Test Application) == Registered custom function 'SHA1' func_sha1.so => (SHA-1 computation dialplan function) == Registered custom function 'BLACKLIST' func_blacklist.so => (Look up Caller*ID name/number from blacklist database) == Parsing '/etc/asterisk/say.conf': == Found == Registered application 'Playback' app_playback.so => (Sound File Playback Application) == 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 '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 'ChannelRedirect' app_channelredirect.so => (Redirects a given channel to a dialplan target) == Registered custom function 'CHANNEL' == Registered custom function 'CHANNELS' == Registered custom function 'MASTER_CHANNEL' func_channel.so => (Channel information dialplan functions) == Registered custom function 'ICONV' func_iconv.so => (Charset conversions) == Registered application 'WaitForRing' app_waitforring.so => (Waits until first ring after time) == Registered custom function 'DB' == Registered custom function 'DB_EXISTS' == Registered custom function 'DB_DELETE' func_db.so => (Database (astdb) related dialplan functions) == Registered application 'ADSIProg' app_adsiprog.so => (Asterisk ADSI Programming Application) == Registered application 'ChanIsAvail' app_chanisavail.so => (Check channel availability) == Registered application 'Authenticate' app_authenticate.so => (Authentication Application) == Registered application 'NBScat' app_nbscat.so => (Silly NBS Stream Application) == Registered application 'DISA' app_disa.so => (DISA (Direct Inward System Access) Application) == Registered application 'Directory' app_directory.so => (Extension Directory) == Registered custom function 'GLOBAL' == Registered custom function 'SHARED' func_global.so => (Variable dialplan functions) == Registered custom function 'RAND' func_rand.so => (Random number dialplan function) == Registered application 'ICES' app_ices.so => (Encode and Stream via icecast and ices) == Registered translator 'gsmtolin' from format gsm to slin, cost 2000 == Registered translator 'lintogsm' from format slin to gsm, cost 5000 codec_gsm.so => (GSM Coder/Decoder) == Registered application 'Page' app_page.so => (Page Multiple Phones) == Registered application 'ControlPlayback' app_controlplayback.so => (Control Playback Application) == 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 'SRVQUERY' == Registered custom function 'SRVRESULT' func_srv.so => (SRV related dialplan functions) == Registered application 'Dictate' app_dictate.so => (Virtual Dictation Machine) == Registered custom function 'DIALPLAN_EXISTS' func_dialplan.so => (Dialplan Context/Extension/Priority Checking Functions) == Registered application 'UserEvent' app_userevent.so => (Custom User Event Application) == Registered application 'CELGenUserEvent' app_celgenuserevent.so => (Generate an User-Defined CEL event) == Registered custom function 'AST_CONFIG' func_config.so => (Asterisk configuration file variable access) == Registered application 'ReadFile' app_readfile.so => (Stores output of file into a variable) == Registered application 'SoftHangup' app_softhangup.so => (Hangs up the requested channel) == Registered custom function 'VOLUME' func_volume.so => (Technology independent volume control) == Registered custom function 'FRAME_TRACE' func_frame_trace.so => (Frame Trace for internal ast_frame debugging.) == Registered custom function 'TIMEOUT' func_timeout.so => (Channel timeout dialplan functions) == AGI Command 'gosub' registered == Registered application 'StackPop' == Registered application 'Return' == Registered application 'GosubIf' == Registered application 'Gosub' == Registered custom function 'LOCAL' == Registered custom function 'LOCAL_PEEK' app_stack.so => (Dialplan subroutines (Gosub, Return, etc)) == Registered translator 'slin16_to_slin8' from format slin16 to slin, cost 101984 == Registered translator 'slin8_to_slin16' from format slin to slin16, cost 8998 codec_resample.so => (SLIN Resampling Codec) == Registered application 'SetCallerPres' app_setcallerid.so => (Set CallerID Presentation Application) == Registered application 'Originate' app_originate.so => (Originate call) == Registered translator 'g726tolin' from format g726 to slin, cost 3000 == Registered translator 'lintog726' from format slin to g726, cost 4999 == Registered translator 'g726aal2tolin' from format g726aal2 to slin, cost 3000 == Registered translator 'lintog726aal2' from format slin to g726aal2, cost 3999 codec_g726.so => (ITU G.726-32kbps G726 Transcoder) == Registered application 'WaitForSilence' == Registered application 'WaitForNoise' app_waitforsilence.so => (Wait For Silence) == Registered custom function 'LOCK' == Registered custom function 'TRYLOCK' == Registered custom function 'UNLOCK' func_lock.so => (Dialplan mutexes) == Registered bridge technology softmix bridge_softmix.so => (Multi-party software based channel mixing) == Registered application 'Milliwatt' app_milliwatt.so => (Digital Milliwatt (mu-law) Test Application) == Registered application 'While' == Registered application 'EndWhile' == Registered application 'ExitWhile' == Registered application 'ContinueWhile' app_while.so => (While Loops and Conditional Execution) == Registered custom function 'ODBC_FETCH' == Registered application 'ODBCFinish' == Parsing '/etc/asterisk/func_odbc.conf': == Found == Registered custom function 'ODBC_SQL' == Registered custom function 'ODBC_ANTIGF' == Registered custom function 'ODBC_PRESENCE' == Registered custom function 'SQL_ESC' func_odbc.so => (ODBC lookups) == Registered custom function 'SHELL' func_shell.so => (Returns the output of a shell command) == Registered application 'Exec' == Registered application 'TryExec' == Registered application 'ExecIf' app_exec.so => (Executes dialplan applications) res_clioriginate.so => (Call origination and redirection from the CLI) == Registered application 'Record' app_record.so => (Trivial Record Application) == Registered application 'NoCDR' app_cdr.so => (Tell Asterisk to not maintain a CDR for the current call) == 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 'SendURL' app_url.so => (Send URL Applications) == Registered custom function 'FIELDQTY' == Registered custom function 'FIELDNUM' == Registered custom function 'FILTER' == Registered custom function 'REPLACE' == Registered custom function 'LISTFILTER' == Registered custom function 'REGEX' == Registered custom function 'ARRAY' == Registered custom function 'QUOTE' == Registered custom function 'CSV_QUOTE' == Registered custom function 'LEN' == Registered custom function 'STRFTIME' == Registered custom function 'STRPTIME' == Registered custom function 'EVAL' == Registered custom function 'KEYPADHASH' == Registered custom function 'HASHKEYS' == Registered custom function 'HASH' == Registered application 'ClearHash' == Registered custom function 'TOUPPER' == Registered custom function 'TOLOWER' == Registered custom function 'SHIFT' == Registered custom function 'POP' == Registered custom function 'PUSH' == Registered custom function 'UNSHIFT' == Registered custom function 'PASSTHRU' func_strings.so => (String handling dialplan functions) == Registered application 'SendText' app_sendtext.so => (Send Text Applications) == Registered application 'SMS' app_sms.so => (SMS/PSTN handler) == Registered translator 'ulawtolin' from format ulaw to slin, cost 1 == Registered translator 'lintoulaw' from format slin to ulaw, cost 1 == Registered translator 'lintotestlaw' from format slin to testlaw, cost 1 == Registered translator 'testlawtolin' from format testlaw to slin, cost 1 codec_ulaw.so => (mu-Law Coder/Decoder) == Registered custom function 'CUT' == Registered custom function 'SORT' func_cut.so => (Cut out information from a string) == 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 'BackgroundDetect' app_talkdetect.so => (Playback with Talk Detection) == Registered custom function 'SYSINFO' func_sysinfo.so => (System information related functions) == Registered application 'Flash' app_flash.so => (Flash channel application) == Registered application 'ExternalIVR' app_externalivr.so => (External IVR Interface Application) == Registered application 'Morsecode' app_morsecode.so => (Morse code) == Registered application 'ParkAndAnnounce' app_parkandannounce.so => (Call Parking and Announce Application) > AlarmReceiver: No config file == Registered custom function 'URIDECODE' == Registered custom function 'URIENCODE' func_uri.so => (URI encode/decode dialplan functions) == Registered application 'TrySystem' == Registered application 'System' app_system.so => (Generic System() application) == Registered application 'Log' == Registered application 'Verbose' app_verbose.so => (Send verbose output) == Registered application 'DAHDIBarge' app_dahdibarge.so => (Barge in on DAHDI channel application) == Parsing '/etc/asterisk/extensions.conf': == Found == Setting global variable 'CONSOLE' to 'Console/dsp' == Setting global variable 'SIP_PM1' to 'SIP/301' == Setting global variable 'SIP_PM2' to 'SIP/302' == Setting global variable 'SIP_PM3' to 'SIP/303' == Setting global variable 'SIP_PM4' to 'SIP/304' == Setting global variable 'SIP_PM5' to 'SIP/305' == Setting global variable 'SOFT_PM1' to 'SIP/401' == Setting global variable 'SOFT_PM2' to 'SIP/402' == Setting global variable 'SOFT_PM3' to 'SIP/403' == Setting global variable 'SOFT_PM4' to 'SIP/404' == Setting global variable 'SOFT_PM5' to 'SIP/405' == Setting global variable 'SOFT_PM6' to 'SIP/406' == Setting global variable 'SOFT_PM7' to 'SIP/407' == Setting global variable 'SOFT_PM8' to 'SIP/408' == Setting global variable 'SOFT_PM9' to 'SIP/409' == Setting global variable 'SOFT_PM10' to 'SIP/410' == Setting global variable 'FXS_PM6' to 'SIP/206' == Setting global variable 'FXS_PM7' to 'SIP/207' == Setting global variable 'FXO_102' to 'SIP/102' == Setting global variable 'FXO_103' to 'SIP/103' == Setting global variable 'TRUNKMSD' to '1' -- Registered extension context 'default'; registrar: pbx_config -- Including context 'phones' in context 'default' -- Including context 'gateways' in context 'default' -- Including context 'iads' in context 'default' -- Registered extension context 'phones'; registrar: pbx_config -- Including context 'labextens' in context 'phones' -- Registered extension context 'gateways'; registrar: pbx_config -- Including context 'labextens' in context 'gateways' -- Registered extension context 'iads'; registrar: pbx_config -- Registered extension context 'international'; registrar: pbx_config -- Including context 'longdistance' in context 'international' -- Including context 'outbound-international' in context 'international' -- Registered extension context 'longdistance'; registrar: pbx_config -- Including context 'localarea' in context 'longdistance' -- Including context 'outbound-longdistance' in context 'longdistance' -- Registered extension context 'localarea'; registrar: pbx_config -- Including context 'outbound-public' in context 'localarea' -- Registered extension context 'internal'; registrar: pbx_config -- Including context 'default' in context 'internal' -- Including context 'outbound-hiik' in context 'internal' -- Including context 'parkedcalls' in context 'internal' -- Registered extension context 'outbound-hiik'; registrar: pbx_config -- Added extension '_7XXX' priority 1 to outbound-hiik -- Added extension '_7XXX' priority 2 to outbound-hiik -- Added extension '_7XXX' priority 3 to outbound-hiik -- Registered extension context 'outbound-public'; registrar: pbx_config -- Added extension '_9NXXXXX' priority 1 to outbound-public -- Added extension '_9NXXXXX' priority 2 to outbound-public -- Added extension '_9NXXXXX' priority 3 to outbound-public -- Registered extension context 'outbound-longdistance'; registrar: pbx_config -- Added extension '_8XXXNXXXXXX' priority 1 to outbound-longdistance -- Added extension '_8XXXNXXXXXX' priority 2 to outbound-longdistance -- Added extension '_8XXXNXXXXXX' priority 3 to outbound-longdistance -- Registered extension context 'outbound-international'; registrar: pbx_config -- Added extension '_810XXXNXXXXXX' priority 1 to outbound-international -- Added extension '_810XXXNXXXXXX' priority 2 to outbound-international -- Added extension '_810XXXNXXXXXX' priority 3 to outbound-international -- Registered extension context 'inbound'; registrar: pbx_config -- Added extension 's' priority 1 to inbound -- Added extension 's' priority 2 to inbound -- Added extension 's' priority 3 to inbound -- Added extension 's' priority 4 to inbound -- Added extension 's' priority 5 to inbound -- Added extension 's' priority 6 to inbound -- Added extension 'i' priority 1 to inbound -- Added extension 'i' priority 2 to inbound -- Added extension 't' priority 1 to inbound -- Added extension 't' priority 2 to inbound -- Registered extension context 'incoming-fxo'; registrar: pbx_config -- Registered extension context 'labextens'; registrar: pbx_config -- Including context 'stdexten' in context 'labextens' -- Added extension '_1XX' priority 1 to labextens -- Added extension '_2XX' priority 1 to labextens -- Added extension '_3XX' priority 1 to labextens -- Added extension '_4XX' priority 1 to labextens -- Registered extension context 'stdexten'; registrar: pbx_config -- Added extension '_X.' priority 50000 to stdexten -- Added extension '_X.' priority 50001 to stdexten -- Added extension '_X.' priority 50002 to stdexten -- Added extension '_X.' priority 50003 to stdexten -- Added extension '_X.' priority 50004 to stdexten -- Added extension '_X.' priority 50005 to stdexten -- Added extension '_X.' priority 50006 to stdexten -- Added extension 'stdexten-NOANSWER' priority 1 to stdexten -- Added extension 'stdexten-NOANSWER' priority 2 to stdexten -- Added extension 'stdexten-NOANSWER' priority 3 to stdexten -- Added extension 'stdexten-BUSY' priority 1 to stdexten -- Added extension 'stdexten-BUSY' priority 2 to stdexten -- Added extension 'stdexten-BUSY' priority 3 to stdexten -- Added extension '_stdexten-.' priority 1 to stdexten -- Added extension 'a' priority 1 to stdexten -- Added extension 'a' priority 2 to stdexten == Parsing '/etc/asterisk/users.conf': == Found -- Registered extension context 'app_dial_gosub_virtual_context'; registrar: app_dial -- merging incls/swits/igpats from old(app_dial_gosub_virtual_context) to new(app_dial_gosub_virtual_context) context, registrar = pbx_config -- Added extension 's' priority 1 to app_dial_gosub_virtual_context -- Registered extension context 'parkedcalls'; registrar: features -- merging incls/swits/igpats from old(parkedcalls) to new(parkedcalls) context, registrar = pbx_config -- Added extension '700' priority 1 to parkedcalls -- Time to scan old dialplan and merge leftovers back into the new: 0.001941 sec -- Time to restore hints and swap in new dialplan: 0.000039 sec -- Time to delete the old dialplan: 0.000017 sec -- Total time merge_contexts_delete: 0.001997 sec pbx_config.so => (Text Extension Configuration) [Feb 19 18:02:52] NOTICE[30811]: chan_ooh323.c:2486 reload_config: Unable to load config ooh323.conf, OOH323 disabled chan_ooh323.so => (Objective Systems H323 Channel) res_limit.so => (Resource limits) == Registered application 'WaitUntil' app_waituntil.so => (Wait until specified time) == Parsing '/etc/asterisk/cli_permissions.conf': == Found Asterisk Ready. == Parsing '/etc/asterisk/cli.conf': == Found *CLI> *CLI> *CLI> sip set debug on SIP Debugging enabled *CLI> rtp set Retransmitting #4 (no NAT) to 192.168.70.201:5060: NOTIFY sip:301@192.168.70.201:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK15079b7b Max-Forwards: 70 From: "asterisk" ;tag=as742f520f To: Contact: Call-ID: 7f99bb724901fb4e30ca372a79db1e15@192.168.70.51:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.8.2.2 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 99 Messages-Waiting: no Message-Account: sip:voicemail@192.168.70.51:5060 Voice-Message: 0/0 (0/0) --- Retransmitting #4 (no NAT) to 192.168.70.202:5060: NOTIFY sip:302@192.168.70.202:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK31c67a7c Max-Forwards: 70 From: "asterisk" ;tag=as3afa37aa To: Contact: Call-ID: 4834d3c80907ad9908a9017431b6d4be@192.168.70.51:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.8.2.2 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 99 Messages-Waiting: no Message-Account: sip:voicemail@192.168.70.51:5060 Voice-Message: 0/0 (0/0) --- Retransmitting #4 (no NAT) to 192.168.70.203:5060: NOTIFY sip:303@192.168.70.203:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK099cf1bd Max-Forwards: 70 From: "asterisk" ;tag=as71f70680 To: Contact: Call-ID: 2d6fe213674431ba2eea3a60697cf904@192.168.70.51:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.8.2.2 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 99 Messages-Waiting: no Message-Account: sip:voicemail@192.168.70.51:5060 Voice-Message: 0/0 (0/0) --- Retransmitting #4 (no NAT) to 192.168.70.204:5060: NOTIFY sip:304@192.168.70.204:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK6d4f7f5b Max-Forwards: 70 From: "asterisk" ;tag=as77610280 To: Contact: Call-ID: 7bbe5e951f68d87c51c78e3f0ae3d0d4@192.168.70.51:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.8.2.2 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 99 Messages-Waiting: no Message-Account: sip:voicemail@192.168.70.51:5060 Voice-Message: 0/0 (0/0) --- Retransmitting #4 (no NAT) to 192.168.70.205:5060: NOTIFY sip:305@192.168.70.205:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK1231fbbe Max-Forwards: 70 From: "asterisk" ;tag=as4aa3d562 To: Contact: Call-ID: 0c55879646d6b0106551d8e554eef861@192.168.70.51:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.8.2.2 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 99 Messages-Waiting: no Message-Account: sip:voicemail@192.168.70.51:5060 Voice-Message: 0/0 (0/0) --- debug on RTP Debugging Enabled *CLI> *CLI> *CLI> Retransmitting #5 (no NAT) to 192.168.70.201:5060: NOTIFY sip:301@192.168.70.201:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK15079b7b Max-Forwards: 70 From: "asterisk" ;tag=as742f520f To: Contact: Call-ID: 7f99bb724901fb4e30ca372a79db1e15@192.168.70.51:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.8.2.2 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 99 Messages-Waiting: no Message-Account: sip:voicemail@192.168.70.51:5060 Voice-Message: 0/0 (0/0) --- Retransmitting #5 (no NAT) to 192.168.70.202:5060: NOTIFY sip:302@192.168.70.202:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK31c67a7c Max-Forwards: 70 From: "asterisk" ;tag=as3afa37aa To: Contact: Call-ID: 4834d3c80907ad9908a9017431b6d4be@192.168.70.51:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.8.2.2 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 99 Messages-Waiting: no Message-Account: sip:voicemail@192.168.70.51:5060 Voice-Message: 0/0 (0/0) --- Retransmitting #5 (no NAT) to 192.168.70.203:5060: NOTIFY sip:303@192.168.70.203:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK099cf1bd Max-Forwards: 70 From: "asterisk" ;tag=as71f70680 To: Contact: Call-ID: 2d6fe213674431ba2eea3a60697cf904@192.168.70.51:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.8.2.2 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 99 Messages-Waiting: no Message-Account: sip:voicemail@192.168.70.51:5060 Voice-Message: 0/0 (0/0) --- Retransmitting #5 (no NAT) to 192.168.70.204:5060: NOTIFY sip:304@192.168.70.204:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK6d4f7f5b Max-Forwards: 70 From: "asterisk" ;tag=as77610280 To: Contact: Call-ID: 7bbe5e951f68d87c51c78e3f0ae3d0d4@192.168.70.51:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.8.2.2 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 99 Messages-Waiting: no Message-Account: sip:voicemail@192.168.70.51:5060 Voice-Message: 0/0 (0/0) --- Retransmitting #5 (no NAT) to 192.168.70.205:5060: NOTIFY sip:305@192.168.70.205:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK1231fbbe Max-Forwards: 70 From: "asterisk" ;tag=as4aa3d562 To: Contact: Call-ID: 0c55879646d6b0106551d8e554eef861@192.168.70.51:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.8.2.2 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 99 Messages-Waiting: no Message-Account: sip:voicemail@192.168.70.51:5060 Voice-Message: 0/0 (0/0) --- Retransmitting #6 (no NAT) to 192.168.70.201:5060: NOTIFY sip:301@192.168.70.201:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK15079b7b Max-Forwards: 70 From: "asterisk" ;tag=as742f520f To: Contact: Call-ID: 7f99bb724901fb4e30ca372a79db1e15@192.168.70.51:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.8.2.2 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 99 Messages-Waiting: no Message-Account: sip:voicemail@192.168.70.51:5060 Voice-Message: 0/0 (0/0) --- Retransmitting #6 (no NAT) to 192.168.70.202:5060: NOTIFY sip:302@192.168.70.202:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK31c67a7c Max-Forwards: 70 From: "asterisk" ;tag=as3afa37aa To: Contact: Call-ID: 4834d3c80907ad9908a9017431b6d4be@192.168.70.51:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.8.2.2 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 99 Messages-Waiting: no Message-Account: sip:voicemail@192.168.70.51:5060 Voice-Message: 0/0 (0/0) --- Retransmitting #6 (no NAT) to 192.168.70.203:5060: NOTIFY sip:303@192.168.70.203:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK099cf1bd Max-Forwards: 70 From: "asterisk" ;tag=as71f70680 To: Contact: Call-ID: 2d6fe213674431ba2eea3a60697cf904@192.168.70.51:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.8.2.2 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 99 Messages-Waiting: no Message-Account: sip:voicemail@192.168.70.51:5060 Voice-Message: 0/0 (0/0) --- Retransmitting #6 (no NAT) to 192.168.70.204:5060: NOTIFY sip:304@192.168.70.204:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK6d4f7f5b Max-Forwards: 70 From: "asterisk" ;tag=as77610280 To: Contact: Call-ID: 7bbe5e951f68d87c51c78e3f0ae3d0d4@192.168.70.51:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.8.2.2 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 99 Messages-Waiting: no Message-Account: sip:voicemail@192.168.70.51:5060 Voice-Message: 0/0 (0/0) --- Retransmitting #6 (no NAT) to 192.168.70.205:5060: NOTIFY sip:305@192.168.70.205:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK1231fbbe Max-Forwards: 70 From: "asterisk" ;tag=as4aa3d562 To: Contact: Call-ID: 0c55879646d6b0106551d8e554eef861@192.168.70.51:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.8.2.2 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 99 Messages-Waiting: no Message-Account: sip:voicemail@192.168.70.51:5060 Voice-Message: 0/0 (0/0) --- Retransmitting #7 (no NAT) to 192.168.70.201:5060: NOTIFY sip:301@192.168.70.201:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK15079b7b Max-Forwards: 70 From: "asterisk" ;tag=as742f520f To: Contact: Call-ID: 7f99bb724901fb4e30ca372a79db1e15@192.168.70.51:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.8.2.2 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 99 Messages-Waiting: no Message-Account: sip:voicemail@192.168.70.51:5060 Voice-Message: 0/0 (0/0) --- Retransmitting #7 (no NAT) to 192.168.70.202:5060: NOTIFY sip:302@192.168.70.202:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK31c67a7c Max-Forwards: 70 From: "asterisk" ;tag=as3afa37aa To: Contact: Call-ID: 4834d3c80907ad9908a9017431b6d4be@192.168.70.51:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.8.2.2 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 99 Messages-Waiting: no Message-Account: sip:voicemail@192.168.70.51:5060 Voice-Message: 0/0 (0/0) --- Retransmitting #7 (no NAT) to 192.168.70.203:5060: NOTIFY sip:303@192.168.70.203:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK099cf1bd Max-Forwards: 70 From: "asterisk" ;tag=as71f70680 To: Contact: Call-ID: 2d6fe213674431ba2eea3a60697cf904@192.168.70.51:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.8.2.2 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 99 Messages-Waiting: no Message-Account: sip:voicemail@192.168.70.51:5060 Voice-Message: 0/0 (0/0) --- Retransmitting #7 (no NAT) to 192.168.70.204:5060: NOTIFY sip:304@192.168.70.204:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK6d4f7f5b Max-Forwards: 70 From: "asterisk" ;tag=as77610280 To: Contact: Call-ID: 7bbe5e951f68d87c51c78e3f0ae3d0d4@192.168.70.51:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.8.2.2 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 99 Messages-Waiting: no Message-Account: sip:voicemail@192.168.70.51:5060 Voice-Message: 0/0 (0/0) --- Retransmitting #7 (no NAT) to 192.168.70.205:5060: NOTIFY sip:305@192.168.70.205:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK1231fbbe Max-Forwards: 70 From: "asterisk" ;tag=as4aa3d562 To: Contact: Call-ID: 0c55879646d6b0106551d8e554eef861@192.168.70.51:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.8.2.2 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 99 Messages-Waiting: no Message-Account: sip:voicemail@192.168.70.51:5060 Voice-Message: 0/0 (0/0) --- Retransmitting #8 (no NAT) to 192.168.70.201:5060: NOTIFY sip:301@192.168.70.201:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK15079b7b Max-Forwards: 70 From: "asterisk" ;tag=as742f520f To: Contact: Call-ID: 7f99bb724901fb4e30ca372a79db1e15@192.168.70.51:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.8.2.2 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 99 Messages-Waiting: no Message-Account: sip:voicemail@192.168.70.51:5060 Voice-Message: 0/0 (0/0) --- Retransmitting #8 (no NAT) to 192.168.70.202:5060: NOTIFY sip:302@192.168.70.202:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK31c67a7c Max-Forwards: 70 From: "asterisk" ;tag=as3afa37aa To: Contact: Call-ID: 4834d3c80907ad9908a9017431b6d4be@192.168.70.51:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.8.2.2 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 99 Messages-Waiting: no Message-Account: sip:voicemail@192.168.70.51:5060 Voice-Message: 0/0 (0/0) --- Retransmitting #8 (no NAT) to 192.168.70.203:5060: NOTIFY sip:303@192.168.70.203:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK099cf1bd Max-Forwards: 70 From: "asterisk" ;tag=as71f70680 To: Contact: Call-ID: 2d6fe213674431ba2eea3a60697cf904@192.168.70.51:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.8.2.2 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 99 Messages-Waiting: no Message-Account: sip:voicemail@192.168.70.51:5060 Voice-Message: 0/0 (0/0) --- Retransmitting #8 (no NAT) to 192.168.70.204:5060: NOTIFY sip:304@192.168.70.204:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK6d4f7f5b Max-Forwards: 70 From: "asterisk" ;tag=as77610280 To: Contact: Call-ID: 7bbe5e951f68d87c51c78e3f0ae3d0d4@192.168.70.51:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.8.2.2 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 99 Messages-Waiting: no Message-Account: sip:voicemail@192.168.70.51:5060 Voice-Message: 0/0 (0/0) --- Retransmitting #8 (no NAT) to 192.168.70.205:5060: NOTIFY sip:305@192.168.70.205:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK1231fbbe Max-Forwards: 70 From: "asterisk" ;tag=as4aa3d562 To: Contact: Call-ID: 0c55879646d6b0106551d8e554eef861@192.168.70.51:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.8.2.2 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 99 Messages-Waiting: no Message-Account: sip:voicemail@192.168.70.51:5060 Voice-Message: 0/0 (0/0) --- Retransmitting #9 (no NAT) to 192.168.70.201:5060: NOTIFY sip:301@192.168.70.201:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK15079b7b Max-Forwards: 70 From: "asterisk" ;tag=as742f520f To: Contact: Call-ID: 7f99bb724901fb4e30ca372a79db1e15@192.168.70.51:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.8.2.2 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 99 Messages-Waiting: no Message-Account: sip:voicemail@192.168.70.51:5060 Voice-Message: 0/0 (0/0) --- Retransmitting #9 (no NAT) to 192.168.70.202:5060: NOTIFY sip:302@192.168.70.202:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK31c67a7c Max-Forwards: 70 From: "asterisk" ;tag=as3afa37aa To: Contact: Call-ID: 4834d3c80907ad9908a9017431b6d4be@192.168.70.51:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.8.2.2 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 99 Messages-Waiting: no Message-Account: sip:voicemail@192.168.70.51:5060 Voice-Message: 0/0 (0/0) --- Retransmitting #9 (no NAT) to 192.168.70.203:5060: NOTIFY sip:303@192.168.70.203:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK099cf1bd Max-Forwards: 70 From: "asterisk" ;tag=as71f70680 To: Contact: Call-ID: 2d6fe213674431ba2eea3a60697cf904@192.168.70.51:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.8.2.2 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 99 Messages-Waiting: no Message-Account: sip:voicemail@192.168.70.51:5060 Voice-Message: 0/0 (0/0) --- Retransmitting #9 (no NAT) to 192.168.70.204:5060: NOTIFY sip:304@192.168.70.204:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK6d4f7f5b Max-Forwards: 70 From: "asterisk" ;tag=as77610280 To: Contact: Call-ID: 7bbe5e951f68d87c51c78e3f0ae3d0d4@192.168.70.51:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.8.2.2 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 99 Messages-Waiting: no Message-Account: sip:voicemail@192.168.70.51:5060 Voice-Message: 0/0 (0/0) --- Retransmitting #9 (no NAT) to 192.168.70.205:5060: NOTIFY sip:305@192.168.70.205:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK1231fbbe Max-Forwards: 70 From: "asterisk" ;tag=as4aa3d562 To: Contact: Call-ID: 0c55879646d6b0106551d8e554eef861@192.168.70.51:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.8.2.2 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 99 Messages-Waiting: no Message-Account: sip:voicemail@192.168.70.51:5060 Voice-Message: 0/0 (0/0) --- Retransmitting #10 (no NAT) to 192.168.70.201:5060: NOTIFY sip:301@192.168.70.201:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK15079b7b Max-Forwards: 70 From: "asterisk" ;tag=as742f520f To: Contact: Call-ID: 7f99bb724901fb4e30ca372a79db1e15@192.168.70.51:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.8.2.2 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 99 Messages-Waiting: no Message-Account: sip:voicemail@192.168.70.51:5060 Voice-Message: 0/0 (0/0) --- Retransmitting #10 (no NAT) to 192.168.70.202:5060: NOTIFY sip:302@192.168.70.202:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK31c67a7c Max-Forwards: 70 From: "asterisk" ;tag=as3afa37aa To: Contact: Call-ID: 4834d3c80907ad9908a9017431b6d4be@192.168.70.51:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.8.2.2 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 99 Messages-Waiting: no Message-Account: sip:voicemail@192.168.70.51:5060 Voice-Message: 0/0 (0/0) --- Retransmitting #10 (no NAT) to 192.168.70.203:5060: NOTIFY sip:303@192.168.70.203:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK099cf1bd Max-Forwards: 70 From: "asterisk" ;tag=as71f70680 To: Contact: Call-ID: 2d6fe213674431ba2eea3a60697cf904@192.168.70.51:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.8.2.2 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 99 Messages-Waiting: no Message-Account: sip:voicemail@192.168.70.51:5060 Voice-Message: 0/0 (0/0) --- Retransmitting #10 (no NAT) to 192.168.70.204:5060: NOTIFY sip:304@192.168.70.204:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK6d4f7f5b Max-Forwards: 70 From: "asterisk" ;tag=as77610280 To: Contact: Call-ID: 7bbe5e951f68d87c51c78e3f0ae3d0d4@192.168.70.51:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.8.2.2 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 99 Messages-Waiting: no Message-Account: sip:voicemail@192.168.70.51:5060 Voice-Message: 0/0 (0/0) --- Retransmitting #10 (no NAT) to 192.168.70.205:5060: NOTIFY sip:305@192.168.70.205:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK1231fbbe Max-Forwards: 70 From: "asterisk" ;tag=as4aa3d562 To: Contact: Call-ID: 0c55879646d6b0106551d8e554eef861@192.168.70.51:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.8.2.2 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 99 Messages-Waiting: no Message-Account: sip:voicemail@192.168.70.51:5060 Voice-Message: 0/0 (0/0) --- [Feb 19 18:03:23] WARNING[30827]: chan_sip.c:3386 retrans_pkt: Retransmission timeout reached on transmission 7f99bb724901fb4e30ca372a79db1e15@192.168.70.51:5060 for seqno 102 (Non-critical Request) -- See doc/sip-retransmit.txt. Packet timed out after 32000ms with no response [Feb 19 18:03:23] WARNING[30827]: chan_sip.c:3386 retrans_pkt: Retransmission timeout reached on transmission 4834d3c80907ad9908a9017431b6d4be@192.168.70.51:5060 for seqno 102 (Non-critical Request) -- See doc/sip-retransmit.txt. Packet timed out after 32000ms with no response [Feb 19 18:03:23] WARNING[30827]: chan_sip.c:3386 retrans_pkt: Retransmission timeout reached on transmission 2d6fe213674431ba2eea3a60697cf904@192.168.70.51:5060 for seqno 102 (Non-critical Request) -- See doc/sip-retransmit.txt. Packet timed out after 32000ms with no response [Feb 19 18:03:23] WARNING[30827]: chan_sip.c:3386 retrans_pkt: Retransmission timeout reached on transmission 7bbe5e951f68d87c51c78e3f0ae3d0d4@192.168.70.51:5060 for seqno 102 (Non-critical Request) -- See doc/sip-retransmit.txt. Packet timed out after 32000ms with no response [Feb 19 18:03:23] WARNING[30827]: chan_sip.c:3386 retrans_pkt: Retransmission timeout reached on transmission 0c55879646d6b0106551d8e554eef861@192.168.70.51:5060 for seqno 102 (Non-critical Request) -- See doc/sip-retransmit.txt. Packet timed out after 32000ms with no response <--- SIP read from UDP:192.168.70.10:25702 ---> REGISTER sip:asterisk.phone.lab403.neis.khabarovsk.su SIP/2.0 Via: SIP/2.0/UDP 192.168.70.10:25702;branch=z9hG4bK-d8754z-125f0f2bdbd318e1-1---d8754z-;rport Max-Forwards: 70 Contact: To: "XLite_PM10" From: "XLite_PM10";tag=cef67e9b Call-ID: ZWViZjdjMWJjZDg4OWY3NDgzNmY4OTBlN2RjMGY5ODY. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite 4 release 4.0 stamp 58832 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 192.168.70.10:25702 (no NAT) <--- Transmitting (no NAT) to 192.168.70.10:25702 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.70.10:25702;branch=z9hG4bK-d8754z-125f0f2bdbd318e1-1---d8754z-;received=192.168.70.10;rport=25702 From: "XLite_PM10";tag=cef67e9b To: "XLite_PM10";tag=as734dcd0d Call-ID: ZWViZjdjMWJjZDg4OWY3NDgzNmY4OTBlN2RjMGY5ODY. CSeq: 1 REGISTER Server: Asterisk PBX 1.8.2.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="phone.lab403.neis.khabarovsk.su", nonce="5e1c7aa9" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'ZWViZjdjMWJjZDg4OWY3NDgzNmY4OTBlN2RjMGY5ODY.' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:192.168.70.10:25702 ---> REGISTER sip:asterisk.phone.lab403.neis.khabarovsk.su SIP/2.0 Via: SIP/2.0/UDP 192.168.70.10:25702;branch=z9hG4bK-d8754z-e1ad8015124b060f-1---d8754z-;rport Max-Forwards: 70 Contact: To: "XLite_PM10" From: "XLite_PM10";tag=cef67e9b Call-ID: ZWViZjdjMWJjZDg4OWY3NDgzNmY4OTBlN2RjMGY5ODY. CSeq: 2 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite 4 release 4.0 stamp 58832 Authorization: Digest username="410",realm="phone.lab403.neis.khabarovsk.su",nonce="5e1c7aa9",uri="sip:asterisk.phone.lab403.neis.khabarovsk.su",response="3f7de113ffe8a9cbfa7de28674846ffb",algorithm=MD5 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 192.168.70.10:25702 (no NAT) -- Registered SIP '410' at 192.168.70.10:25702 > Saved useragent "X-Lite 4 release 4.0 stamp 58832" for peer 410 <--- Transmitting (no NAT) to 192.168.70.10:25702 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.10:25702;branch=z9hG4bK-d8754z-e1ad8015124b060f-1---d8754z-;received=192.168.70.10;rport=25702 From: "XLite_PM10";tag=cef67e9b To: "XLite_PM10";tag=as734dcd0d Call-ID: ZWViZjdjMWJjZDg4OWY3NDgzNmY4OTBlN2RjMGY5ODY. CSeq: 2 REGISTER Server: Asterisk PBX 1.8.2.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 3600 Contact: ;expires=3600 Date: Sat, 19 Feb 2011 08:03:42 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'ZWViZjdjMWJjZDg4OWY3NDgzNmY4OTBlN2RjMGY5ODY.' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:192.168.70.10:25702 ---> SUBSCRIBE sip:410@asterisk.phone.lab403.neis.khabarovsk.su SIP/2.0 Via: SIP/2.0/UDP 192.168.70.10:25702;branch=z9hG4bK-d8754z-d5e419ec600ea227-1---d8754z-;rport Max-Forwards: 70 Contact: To: "XLite_PM10" From: "XLite_PM10";tag=c545cdaf Call-ID: M2E4NzgzYmQxNzQ3ZDFkMmY3YWU0ZTU1OGJkMTFhZTE. CSeq: 1 SUBSCRIBE Expires: 300 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite 4 release 4.0 stamp 58832 Event: message-summary Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Creating new subscription Sending to 192.168.70.10:25702 (no NAT) list_route: hop: Found peer '410' for '410' from 192.168.70.10:25702 <--- Transmitting (no NAT) to 192.168.70.10:25702 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.70.10:25702;branch=z9hG4bK-d8754z-d5e419ec600ea227-1---d8754z-;received=192.168.70.10;rport=25702 From: "XLite_PM10";tag=c545cdaf To: "XLite_PM10";tag=as626c2f5d Call-ID: M2E4NzgzYmQxNzQ3ZDFkMmY3YWU0ZTU1OGJkMTFhZTE. CSeq: 1 SUBSCRIBE Server: Asterisk PBX 1.8.2.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="phone.lab403.neis.khabarovsk.su", nonce="5645fa98" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'M2E4NzgzYmQxNzQ3ZDFkMmY3YWU0ZTU1OGJkMTFhZTE.' in 32000 ms (Method: SUBSCRIBE) <--- SIP read from UDP:192.168.70.10:25702 ---> SUBSCRIBE sip:410@asterisk.phone.lab403.neis.khabarovsk.su SIP/2.0 Via: SIP/2.0/UDP 192.168.70.10:25702;branch=z9hG4bK-d8754z-e56cce572dd3c843-1---d8754z-;rport Max-Forwards: 70 Contact: To: "XLite_PM10" From: "XLite_PM10";tag=c545cdaf Call-ID: M2E4NzgzYmQxNzQ3ZDFkMmY3YWU0ZTU1OGJkMTFhZTE. CSeq: 2 SUBSCRIBE Expires: 300 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite 4 release 4.0 stamp 58832 Authorization: Digest username="410",realm="phone.lab403.neis.khabarovsk.su",nonce="5645fa98",uri="sip:410@asterisk.phone.lab403.neis.khabarovsk.su",response="5e46505af2cc39a285822e80cd252dfb",algorithm=MD5 Event: message-summary Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Creating new subscription Sending to 192.168.70.10:25702 (no NAT) Found peer '410' for '410' from 192.168.70.10:25702 Scheduling destruction of SIP dialog 'M2E4NzgzYmQxNzQ3ZDFkMmY3YWU0ZTU1OGJkMTFhZTE.' in 310000 ms (Method: SUBSCRIBE) <--- Transmitting (no NAT) to 192.168.70.10:25702 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.10:25702;branch=z9hG4bK-d8754z-e56cce572dd3c843-1---d8754z-;received=192.168.70.10;rport=25702 From: "XLite_PM10";tag=c545cdaf To: "XLite_PM10";tag=as626c2f5d Call-ID: M2E4NzgzYmQxNzQ3ZDFkMmY3YWU0ZTU1OGJkMTFhZTE. CSeq: 2 SUBSCRIBE Server: Asterisk PBX 1.8.2.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 300 Contact: ;expires=300 Content-Length: 0 <------------> Reliably Transmitting (no NAT) to 192.168.70.10:25702: NOTIFY sip:410@192.168.70.10:25702 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK0d37167c Max-Forwards: 70 Route: From: "asterisk" ;tag=as626c2f5d To: ;tag=c545cdaf Contact: Call-ID: M2E4NzgzYmQxNzQ3ZDFkMmY3YWU0ZTU1OGJkMTFhZTE. CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.8.2.2 Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 99 Messages-Waiting: no Message-Account: sip:voicemail@192.168.70.51:5060 Voice-Message: 0/0 (0/0) --- <--- SIP read from UDP:192.168.70.10:25702 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK0d37167c Contact: To: ;tag=c545cdaf From: "asterisk";tag=as626c2f5d Call-ID: M2E4NzgzYmQxNzQ3ZDFkMmY3YWU0ZTU1OGJkMTFhZTE. CSeq: 102 NOTIFY User-Agent: X-Lite 4 release 4.0 stamp 58832 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:192.168.70.9:38628 ---> REGISTER sip:asterisk.phone.lab403.neis.khabarovsk.su SIP/2.0 Via: SIP/2.0/UDP 192.168.70.9:38628;branch=z9hG4bK-d8754z-1bd4f8ff5f2b7682-1---d8754z-;rport Max-Forwards: 70 Contact: To: "XLite_PM9" From: "XLite_PM9";tag=8c320d6d Call-ID: ZGQ4Y2Y0ODFkZDk4MWY0Nzc3ODJmOTI3NGFhYjZmN2E. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite 4 release 4.0 stamp 58832 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 192.168.70.9:38628 (no NAT) <--- Transmitting (no NAT) to 192.168.70.9:38628 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.70.9:38628;branch=z9hG4bK-d8754z-1bd4f8ff5f2b7682-1---d8754z-;received=192.168.70.9;rport=38628 From: "XLite_PM9";tag=8c320d6d To: "XLite_PM9";tag=as07b6c32f Call-ID: ZGQ4Y2Y0ODFkZDk4MWY0Nzc3ODJmOTI3NGFhYjZmN2E. CSeq: 1 REGISTER Server: Asterisk PBX 1.8.2.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="phone.lab403.neis.khabarovsk.su", nonce="16c2ed69" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'ZGQ4Y2Y0ODFkZDk4MWY0Nzc3ODJmOTI3NGFhYjZmN2E.' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:192.168.70.9:38628 ---> REGISTER sip:asterisk.phone.lab403.neis.khabarovsk.su SIP/2.0 Via: SIP/2.0/UDP 192.168.70.9:38628;branch=z9hG4bK-d8754z-ba33d2616d18598c-1---d8754z-;rport Max-Forwards: 70 Contact: To: "XLite_PM9" From: "XLite_PM9";tag=8c320d6d Call-ID: ZGQ4Y2Y0ODFkZDk4MWY0Nzc3ODJmOTI3NGFhYjZmN2E. CSeq: 2 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite 4 release 4.0 stamp 58832 Authorization: Digest username="409",realm="phone.lab403.neis.khabarovsk.su",nonce="16c2ed69",uri="sip:asterisk.phone.lab403.neis.khabarovsk.su",response="f41c3cde46ae13fb4f705a50dafd6d74",algorithm=MD5 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 192.168.70.9:38628 (no NAT) -- Registered SIP '409' at 192.168.70.9:38628 > Saved useragent "X-Lite 4 release 4.0 stamp 58832" for peer 409 <--- Transmitting (no NAT) to 192.168.70.9:38628 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.9:38628;branch=z9hG4bK-d8754z-ba33d2616d18598c-1---d8754z-;received=192.168.70.9;rport=38628 From: "XLite_PM9";tag=8c320d6d To: "XLite_PM9";tag=as07b6c32f Call-ID: ZGQ4Y2Y0ODFkZDk4MWY0Nzc3ODJmOTI3NGFhYjZmN2E. CSeq: 2 REGISTER Server: Asterisk PBX 1.8.2.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 3600 Contact: ;expires=3600 Date: Sat, 19 Feb 2011 08:03:54 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'ZGQ4Y2Y0ODFkZDk4MWY0Nzc3ODJmOTI3NGFhYjZmN2E.' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:192.168.70.9:38628 ---> SUBSCRIBE sip:409@asterisk.phone.lab403.neis.khabarovsk.su SIP/2.0 Via: SIP/2.0/UDP 192.168.70.9:38628;branch=z9hG4bK-d8754z-a21560a60e82acab-1---d8754z-;rport Max-Forwards: 70 Contact: To: "XLite_PM9" From: "XLite_PM9";tag=cb5b8854 Call-ID: MmRiODA1MDMyMmZjNjljYWE2YmVjODJkMDZkZWM1YzQ. CSeq: 1 SUBSCRIBE Expires: 300 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite 4 release 4.0 stamp 58832 Event: message-summary Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Creating new subscription Sending to 192.168.70.9:38628 (no NAT) list_route: hop: Found peer '409' for '409' from 192.168.70.9:38628 <--- Transmitting (no NAT) to 192.168.70.9:38628 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.70.9:38628;branch=z9hG4bK-d8754z-a21560a60e82acab-1---d8754z-;received=192.168.70.9;rport=38628 From: "XLite_PM9";tag=cb5b8854 To: "XLite_PM9";tag=as2e326dd9 Call-ID: MmRiODA1MDMyMmZjNjljYWE2YmVjODJkMDZkZWM1YzQ. CSeq: 1 SUBSCRIBE Server: Asterisk PBX 1.8.2.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="phone.lab403.neis.khabarovsk.su", nonce="487ce704" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'MmRiODA1MDMyMmZjNjljYWE2YmVjODJkMDZkZWM1YzQ.' in 32000 ms (Method: SUBSCRIBE) <--- SIP read from UDP:192.168.70.9:38628 ---> SUBSCRIBE sip:409@asterisk.phone.lab403.neis.khabarovsk.su SIP/2.0 Via: SIP/2.0/UDP 192.168.70.9:38628;branch=z9hG4bK-d8754z-3fd2e42934a8bb38-1---d8754z-;rport Max-Forwards: 70 Contact: To: "XLite_PM9" From: "XLite_PM9";tag=cb5b8854 Call-ID: MmRiODA1MDMyMmZjNjljYWE2YmVjODJkMDZkZWM1YzQ. CSeq: 2 SUBSCRIBE Expires: 300 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite 4 release 4.0 stamp 58832 Authorization: Digest username="409",realm="phone.lab403.neis.khabarovsk.su",nonce="487ce704",uri="sip:409@asterisk.phone.lab403.neis.khabarovsk.su",response="0e3bf6e3648d321d09adb1657302c0b2",algorithm=MD5 Event: message-summary Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Creating new subscription Sending to 192.168.70.9:38628 (no NAT) Found peer '409' for '409' from 192.168.70.9:38628 Scheduling destruction of SIP dialog 'MmRiODA1MDMyMmZjNjljYWE2YmVjODJkMDZkZWM1YzQ.' in 310000 ms (Method: SUBSCRIBE) <--- Transmitting (no NAT) to 192.168.70.9:38628 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.9:38628;branch=z9hG4bK-d8754z-3fd2e42934a8bb38-1---d8754z-;received=192.168.70.9;rport=38628 From: "XLite_PM9";tag=cb5b8854 To: "XLite_PM9";tag=as2e326dd9 Call-ID: MmRiODA1MDMyMmZjNjljYWE2YmVjODJkMDZkZWM1YzQ. CSeq: 2 SUBSCRIBE Server: Asterisk PBX 1.8.2.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 300 Contact: ;expires=300 Content-Length: 0 <------------> Reliably Transmitting (no NAT) to 192.168.70.9:38628: NOTIFY sip:409@192.168.70.9:38628 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK3d31af75 Max-Forwards: 70 Route: From: "asterisk" ;tag=as2e326dd9 To: ;tag=cb5b8854 Contact: Call-ID: MmRiODA1MDMyMmZjNjljYWE2YmVjODJkMDZkZWM1YzQ. CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.8.2.2 Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 99 Messages-Waiting: no Message-Account: sip:voicemail@192.168.70.51:5060 Voice-Message: 0/0 (0/0) --- <--- SIP read from UDP:192.168.70.9:38628 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK3d31af75 Contact: To: ;tag=cb5b8854 From: "asterisk";tag=as2e326dd9 Call-ID: MmRiODA1MDMyMmZjNjljYWE2YmVjODJkMDZkZWM1YzQ. CSeq: 102 NOTIFY User-Agent: X-Lite 4 release 4.0 stamp 58832 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '7f99bb724901fb4e30ca372a79db1e15@192.168.70.51:5060' Method: NOTIFY Really destroying SIP dialog '4834d3c80907ad9908a9017431b6d4be@192.168.70.51:5060' Method: NOTIFY Really destroying SIP dialog '2d6fe213674431ba2eea3a60697cf904@192.168.70.51:5060' Method: NOTIFY Really destroying SIP dialog '7bbe5e951f68d87c51c78e3f0ae3d0d4@192.168.70.51:5060' Method: NOTIFY Really destroying SIP dialog '0c55879646d6b0106551d8e554eef861@192.168.70.51:5060' Method: NOTIFY *CLI> *CLI> Really destroying SIP dialog 'ZWViZjdjMWJjZDg4OWY3NDgzNmY4OTBlN2RjMGY5ODY.' Method: REGISTER Really destroying SIP dialog 'ZGQ4Y2Y0ODFkZDk4MWY0Nzc3ODJmOTI3NGFhYjZmN2E.' Method: REGISTER *CLI> *CLI> sip show settings Global Settings: ---------------- UDP Bindaddress: 192.168.70.51:5060 TCP SIP Bindaddress: Disabled TLS SIP Bindaddress: Disabled Videosupport: Yes Textsupport: No Ignore SDP sess. ver.: No AutoCreate Peer: No Match Auth Username: Yes Allow unknown access: No Allow subscriptions: Yes Allow overlap dialing: Yes Allow promsic. redir: No Enable call counters: Yes SIP domain support: Yes Realm. auth: No Our auth realm phone.lab403.neis.khabarovsk.su Use domains as realms: No Call to non-local dom.: No URI user is phone no: No Always auth rejects: Yes Direct RTP setup: No User Agent: Asterisk PBX 1.8.2.2 SDP Session Name: Asterisk PBX 1.8.2.2 SDP Owner Name: root Reg. context: (not set) Regexten on Qualify: No Caller ID: asterisk From: Domain: Record SIP history: Off Call Events: On Auth. Failure Events: On T.38 support: Yes T.38 EC mode: FEC T.38 MaxDtgrm: -1 SIP realtime: Disabled Qualify Freq : 60000 ms Q.850 Reason header: No Network QoS Settings: --------------------------- IP ToS SIP: AF31 IP ToS RTP audio: EF IP ToS RTP video: AF41 IP ToS RTP text: AF21 802.1p CoS SIP: 3 802.1p CoS RTP audio: 5 802.1p CoS RTP video: 4 802.1p CoS RTP text: 2 Jitterbuffer enabled: No Jitterbuffer forced: No Jitterbuffer max size: -1 Jitterbuffer resync: -1 Jitterbuffer impl: Jitterbuffer log: No Network Settings: --------------------------- SIP address remapping: Enabled using externaddr Externhost: externaddr: 192.168.2.9:0 Externrefresh: 10 Localnet: 192.168.50.0/255.255.255.0 192.168.70.0/255.255.255.0 Global Signalling Settings: --------------------------- Codecs: 0x80000008000e (gsm|ulaw|alaw|h263|testlaw) Codec Order: none Relax DTMF: No RFC2833 Compensation: No Symmetric RTP: No Compact SIP headers: No RTP Keepalive: 0 (Disabled) RTP Timeout: 0 (Disabled) RTP Hold Timeout: 0 (Disabled) MWI NOTIFY mime type: application/simple-message-summary DNS SRV lookup: Yes Pedantic SIP support: Yes Reg. min duration 60 secs Reg. max duration: 3600 secs Reg. default duration: 120 secs Outbound reg. timeout: 20 secs Outbound reg. attempts: 0 Notify ringing state: Yes Include CID: No Notify hold state: No SIP Transfer mode: open Max Call Bitrate: 8192 kbps Auto-Framing: No Outb. proxy: Session Timers: Accept Session Refresher: uas Session Expires: 1800 secs Session Min-SE: 90 secs Timer T1: 500 Timer T1 minimum: 100 Timer B: 32000 No premature media: Yes Max forwards: 70 Default Settings: ----------------- Allowed transports: UDP Outbound transport: UDP Context: default Force rport: No DTMF: rfc2833 Qualify: 0 Use ClientCode: No Progress inband: Never Language: ru MOH Interpret: default MOH Suggest: default Voice Mail Extension: asterisk ---- *CLI> <--- SIP read from UDP:192.168.70.10:25702 ---> INVITE sip:409@asterisk.phone.lab403.neis.khabarovsk.su SIP/2.0 Via: SIP/2.0/UDP 192.168.70.10:25702;branch=z9hG4bK-d8754z-a03f301c5a7af773-1---d8754z-;rport Max-Forwards: 70 Contact: To: From: "XLite_PM10";tag=ac93dede Call-ID: MGYyNzdhMDAyYzE0NzZlMDFjMmQ1MTcyYzdlM2I1N2Y. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite 4 release 4.0 stamp 58832 Content-Length: 709 v=0 o=- 12942576345423565 1 IN IP4 192.168.70.10 s=CounterPath X-Lite 4.0 c=IN IP4 192.168.70.10 t=0 0 a=ice-ufrag:eae5fa a=ice-pwd:a3d299009f9a92709e8403171a72aa77 m=audio 55966 RTP/AVP 107 6 98 8 3 101 a=rtpmap:107 BV32/16000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.10 55966 typ host a=candidate:1 2 UDP 659134 192.168.70.10 55967 typ host m=video 62960 RTP/AVP 34 115 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=2;CIF=3 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=3;I=1;J=1;T=1 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.10 62960 typ host a=candidate:1 2 UDP 659134 192.168.70.10 62961 typ host <-------------> --- (13 headers 23 lines) --- == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 Sending to 192.168.70.10:25702 (no NAT) Using INVITE request as basis request - MGYyNzdhMDAyYzE0NzZlMDFjMmQ1MTcyYzdlM2I1N2Y. Found peer '410' for '410' from 192.168.70.10:25702 <--- Reliably Transmitting (no NAT) to 192.168.70.10:25702 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.70.10:25702;branch=z9hG4bK-d8754z-a03f301c5a7af773-1---d8754z-;received=192.168.70.10;rport=25702 From: "XLite_PM10";tag=ac93dede To: ;tag=as32a57108 Call-ID: MGYyNzdhMDAyYzE0NzZlMDFjMmQ1MTcyYzdlM2I1N2Y. CSeq: 1 INVITE Server: Asterisk PBX 1.8.2.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="phone.lab403.neis.khabarovsk.su", nonce="0e980c3c" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'MGYyNzdhMDAyYzE0NzZlMDFjMmQ1MTcyYzdlM2I1N2Y.' in 32000 ms (Method: INVITE) <--- SIP read from UDP:192.168.70.10:25702 ---> ACK sip:409@asterisk.phone.lab403.neis.khabarovsk.su SIP/2.0 Via: SIP/2.0/UDP 192.168.70.10:25702;branch=z9hG4bK-d8754z-a03f301c5a7af773-1---d8754z-;rport Max-Forwards: 70 To: ;tag=as32a57108 From: "XLite_PM10";tag=ac93dede Call-ID: MGYyNzdhMDAyYzE0NzZlMDFjMmQ1MTcyYzdlM2I1N2Y. CSeq: 1 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:192.168.70.10:25702 ---> INVITE sip:409@asterisk.phone.lab403.neis.khabarovsk.su SIP/2.0 Via: SIP/2.0/UDP 192.168.70.10:25702;branch=z9hG4bK-d8754z-f29cfcbc0ed99e81-1---d8754z-;rport Max-Forwards: 70 Contact: To: From: "XLite_PM10";tag=ac93dede Call-ID: MGYyNzdhMDAyYzE0NzZlMDFjMmQ1MTcyYzdlM2I1N2Y. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite 4 release 4.0 stamp 58832 Authorization: Digest username="410",realm="phone.lab403.neis.khabarovsk.su",nonce="0e980c3c",uri="sip:409@asterisk.phone.lab403.neis.khabarovsk.su",response="af953fb6c6e590da94f825e0881aaa56",algorithm=MD5 Content-Length: 709 v=0 o=- 12942576345423565 1 IN IP4 192.168.70.10 s=CounterPath X-Lite 4.0 c=IN IP4 192.168.70.10 t=0 0 a=ice-ufrag:eae5fa a=ice-pwd:a3d299009f9a92709e8403171a72aa77 m=audio 55966 RTP/AVP 107 6 98 8 3 101 a=rtpmap:107 BV32/16000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.10 55966 typ host a=candidate:1 2 UDP 659134 192.168.70.10 55967 typ host m=video 62960 RTP/AVP 34 115 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=2;CIF=3 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=3;I=1;J=1;T=1 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.10 62960 typ host a=candidate:1 2 UDP 659134 192.168.70.10 62961 typ host <-------------> --- (14 headers 23 lines) --- Sending to 192.168.70.10:25702 (no NAT) Using INVITE request as basis request - MGYyNzdhMDAyYzE0NzZlMDFjMmQ1MTcyYzdlM2I1N2Y. Found peer '410' for '410' from 192.168.70.10:25702 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Found RTP audio format 107 Found RTP audio format 6 Found RTP audio format 98 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 101 Found audio description format BV32 for ID 107 Found audio description format iLBC for ID 98 Found audio description format telephone-event for ID 101 Found RTP video format 34 Found RTP video format 115 Found video description format H263 for ID 34 Found video description format H263-1998 for ID 115 Capabilities: us - 0x3c078b (g723|gsm|alaw|lpc10|g729|speex|ilbc|h261|h263|h263p|h264), peer - audio=0x42a (gsm|alaw|adpcm|ilbc)/video=0x180000 (h263|h263p)/text=0x0 (nothing), combined - 0x18040a (gsm|alaw|ilbc|h263|h263p) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.70.10:55966 Peer video RTP is at port 192.168.70.10:62960 Peer doesn't provide T.140 Looking for 409 in phones (domain asterisk.phone.lab403.neis.khabarovsk.su) list_route: hop: <--- Transmitting (no NAT) to 192.168.70.10:25702 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.70.10:25702;branch=z9hG4bK-d8754z-f29cfcbc0ed99e81-1---d8754z-;received=192.168.70.10;rport=25702 From: "XLite_PM10";tag=ac93dede To: Call-ID: MGYyNzdhMDAyYzE0NzZlMDFjMmQ1MTcyYzdlM2I1N2Y. CSeq: 2 INVITE Server: Asterisk PBX 1.8.2.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [409@phones:1] Gosub("SIP/410-00000000", "409,stdexten(SIP/409)") in new stack -- Executing [409@phones:50000] NoOp("SIP/410-00000000", "Start stdexten") in new stack -- Executing [409@phones:50001] Set("SIP/410-00000000", "LOCAL(ext)=409") in new stack -- Executing [409@phones:50002] Set("SIP/410-00000000", "LOCAL(dev)=SIP/409") in new stack -- Executing [409@phones:50003] Set("SIP/410-00000000", "LOCAL(cntx)=") in new stack -- Executing [409@phones:50004] Set("SIP/410-00000000", "LOCAL(mbx)="409"""") in new stack -- Executing [409@phones:50005] Dial("SIP/410-00000000", "SIP/409,30") in new stack == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Audio is at 5060 Video is at 192.168.70.51:5060 Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x80 (lpc10) to SDP Adding video codec 0x80000 (h263) to SDP Adding video codec 0x100000 (h263p) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.70.9:38628: INVITE sip:409@192.168.70.9:38628;rinstance=51c54eb541dc3435 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK6e228607 Max-Forwards: 70 From: "PM10 Soft-Phone" ;tag=as5c51a7b5 To: Contact: Call-ID: 738ccdde7c12dfbd13a7f0fa3e4a422f@192.168.70.51:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.2.2 Date: Sat, 19 Feb 2011 08:05:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Remote-Party-ID: "PM10 Soft-Phone" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 387 v=0 o=root 692438885 692438885 IN IP4 192.168.70.51 s=Asterisk PBX 1.8.2.2 c=IN IP4 192.168.70.51 b=CT:8192 t=0 0 m=audio 18966 RTP/AVP 8 3 7 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:7 LPC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 19764 RTP/AVP 34 98 a=rtpmap:34 H263/90000 a=rtpmap:98 h263-1998/90000 a=sendrecv --- -- Called 409 <--- SIP read from UDP:192.168.70.9:38628 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK6e228607 To: From: "PM10 Soft-Phone" ;tag=as5c51a7b5 Call-ID: 738ccdde7c12dfbd13a7f0fa3e4a422f@192.168.70.51:5060 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from UDP:192.168.70.9:38628 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK6e228607 Contact: To: ;tag=0f5a5926 From: "PM10 Soft-Phone";tag=as5c51a7b5 Call-ID: 738ccdde7c12dfbd13a7f0fa3e4a422f@192.168.70.51:5060 CSeq: 102 INVITE User-Agent: X-Lite 4 release 4.0 stamp 58832 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- -- SIP/409-00000001 is ringing <--- Transmitting (no NAT) to 192.168.70.10:25702 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.70.10:25702;branch=z9hG4bK-d8754z-f29cfcbc0ed99e81-1---d8754z-;received=192.168.70.10;rport=25702 From: "XLite_PM10";tag=ac93dede To: ;tag=as20e0c424 Call-ID: MGYyNzdhMDAyYzE0NzZlMDFjMmQ1MTcyYzdlM2I1N2Y. CSeq: 2 INVITE Server: Asterisk PBX 1.8.2.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> Got RTP packet from 192.168.70.9:63638 (type 08, seq 004980, ts 3085300, len 000160) Got RTP packet from 192.168.70.9:63638 (type 08, seq 004981, ts 3085460, len 000160) Got RTP packet from 192.168.70.9:63638 (type 08, seq 004982, ts 3085620, len 000160) <--- SIP read from UDP:192.168.70.9:38628 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK6e228607 Contact: To: ;tag=0f5a5926 From: "PM10 Soft-Phone";tag=as5c51a7b5 Call-ID: 738ccdde7c12dfbd13a7f0fa3e4a422f@192.168.70.51:5060 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite 4 release 4.0 stamp 58832 Content-Length: 655 v=0 o=- 12942576351816681 1 IN IP4 192.168.70.9 s=CounterPath X-Lite 4.0 c=IN IP4 192.168.70.9 t=0 0 a=ice-ufrag:9fb215 a=ice-pwd:6e9ec541ff57f81d53879f0807748252 m=audio 63638 RTP/AVP 8 3 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.9 63638 typ host a=candidate:1 2 UDP 659134 192.168.70.9 63639 typ host m=video 49624 RTP/AVP 34 98 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=1;CIF=1;VGA=1 a=rtpmap:98 H263-1998/90000 a=fmtp:98 QCIF=1;CIF=1;VGA=1;I=1;J=1;T=1 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.9 49624 typ host a=candidate:1 2 UDP 659134 192.168.70.9 49625 typ host <-------------> --- (12 headers 21 lines) --- Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 101 Found audio description format telephone-event for ID 101 Found RTP video format 34 Found RTP video format 98 Found video description format H263 for ID 34 Found video description format H263-1998 for ID 98 Capabilities: us - 0x3c078b (g723|gsm|alaw|lpc10|g729|speex|ilbc|h261|h263|h263p|h264), peer - audio=0xa (gsm|alaw)/video=0x180000 (h263|h263p)/text=0x0 (nothing), combined - 0x18000a (gsm|alaw|h263|h263p) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.70.9:63638 Peer video RTP is at port 192.168.70.9:49624 Peer doesn't provide T.140 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.70.9:38628 Transmitting (no NAT) to 192.168.70.9:38628: ACK sip:409@192.168.70.9:38628;rinstance=51c54eb541dc3435 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK39d8dc0d Max-Forwards: 70 From: "PM10 Soft-Phone" ;tag=as5c51a7b5 To: ;tag=0f5a5926 Contact: Call-ID: 738ccdde7c12dfbd13a7f0fa3e4a422f@192.168.70.51:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.2.2 Content-Length: 0 --- -- SIP/409-00000001 answered SIP/410-00000000 Audio is at 5060 Video is at 192.168.70.51:5060 Adding codec 0x8 (alaw) to SDP Adding codec 0x400 (ilbc) to SDP Adding codec 0x2 (gsm) to SDP Adding video codec 0x80000 (h263) to SDP Adding video codec 0x100000 (h263p) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 192.168.70.10:25702 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.10:25702;branch=z9hG4bK-d8754z-f29cfcbc0ed99e81-1---d8754z-;received=192.168.70.10;rport=25702 From: "XLite_PM10";tag=ac93dede To: ;tag=as20e0c424 Call-ID: MGYyNzdhMDAyYzE0NzZlMDFjMmQ1MTcyYzdlM2I1N2Y. CSeq: 2 INVITE Server: Asterisk PBX 1.8.2.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Remote-Party-ID: "PM9 Soft-Phone" ;party=called;privacy=off;screen=no Content-Type: application/sdp Content-Length: 413 v=0 o=root 1789630608 1789630608 IN IP4 192.168.70.51 s=Asterisk PBX 1.8.2.2 c=IN IP4 192.168.70.51 b=CT:8192 t=0 0 m=audio 19650 RTP/AVP 8 98 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 13610 RTP/AVP 34 115 a=rtpmap:34 H263/90000 a=rtpmap:115 h263-1998/90000 a=sendrecv <------------> -- Remotely bridging SIP/410-00000000 and SIP/409-00000001 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.70.9:38628 Audio is at 5060 Video is at 192.168.70.51:5060 Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding video codec 0x80000 (h263) to SDP Adding video codec 0x100000 (h263p) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.70.9:38628: INVITE sip:409@192.168.70.9:38628;rinstance=51c54eb541dc3435 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK288967b6 Max-Forwards: 70 From: "PM10 Soft-Phone" ;tag=as5c51a7b5 To: ;tag=0f5a5926 Contact: Call-ID: 738ccdde7c12dfbd13a7f0fa3e4a422f@192.168.70.51:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.8.2.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Remote-Party-ID: "PM10 Soft-Phone" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 364 v=0 o=root 692438885 692438886 IN IP4 192.168.70.10 s=Asterisk PBX 1.8.2.2 c=IN IP4 192.168.70.10 b=CT:8192 t=0 0 m=audio 55966 RTP/AVP 8 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 62960 RTP/AVP 34 98 a=rtpmap:34 H263/90000 a=rtpmap:98 h263-1998/90000 a=sendrecv --- Sent RTP P2P packet to 192.168.70.10:55966 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.10:55966 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.10:55966 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.10:55966 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.10:55966 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.10:55966 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.10:55966 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.10:55966 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.10:55966 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:63638 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:63638 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:63638 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:63638 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:63638 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:63638 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:63638 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:63638 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:63638 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:63638 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:63638 (type 08, len 000160) <--- SIP read from UDP:192.168.70.9:38628 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK288967b6 Contact: To: ;tag=0f5a5926 From: "PM10 Soft-Phone";tag=as5c51a7b5 Call-ID: 738ccdde7c12dfbd13a7f0fa3e4a422f@192.168.70.51:5060 CSeq: 103 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite 4 release 4.0 stamp 58832 Content-Length: 655 v=0 o=- 12942576351816681 2 IN IP4 192.168.70.9 s=CounterPath X-Lite 4.0 c=IN IP4 192.168.70.9 t=0 0 a=ice-ufrag:9fb215 a=ice-pwd:6e9ec541ff57f81d53879f0807748252 m=audio 63638 RTP/AVP 8 3 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.9 63638 typ host a=candidate:1 2 UDP 659134 192.168.70.9 63639 typ host m=video 49624 RTP/AVP 34 98 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=1;CIF=1;VGA=1 a=rtpmap:98 H263-1998/90000 a=fmtp:98 QCIF=1;CIF=1;VGA=1;I=1;J=1;T=1 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.9 49624 typ host a=candidate:1 2 UDP 659134 192.168.70.9 49625 typ host <-------------> --- (12 headers 21 lines) --- Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 101 Found audio description format telephone-event for ID 101 Found RTP video format 34 Found RTP video format 98 Found video description format H263 for ID 34 Found video description format H263-1998 for ID 98 Capabilities: us - 0x3c078b (g723|gsm|alaw|lpc10|g729|speex|ilbc|h261|h263|h263p|h264), peer - audio=0xa (gsm|alaw)/video=0x180000 (h263|h263p)/text=0x0 (nothing), combined - 0x18000a (gsm|alaw|h263|h263p) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.70.9:63638 Peer video RTP is at port 192.168.70.9:49624 Peer doesn't provide T.140 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.70.9:38628 Transmitting (no NAT) to 192.168.70.9:38628: ACK sip:409@192.168.70.9:38628;rinstance=51c54eb541dc3435 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK6552b382 Max-Forwards: 70 From: "PM10 Soft-Phone" ;tag=as5c51a7b5 To: ;tag=0f5a5926 Contact: Call-ID: 738ccdde7c12dfbd13a7f0fa3e4a422f@192.168.70.51:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 1.8.2.2 Content-Length: 0 --- Sent RTP P2P packet to 192.168.70.9:63638 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:63638 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:63638 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:63638 (type 08, len 000160) Retransmitting #1 (no NAT) to 192.168.70.10:25702: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.10:25702;branch=z9hG4bK-d8754z-f29cfcbc0ed99e81-1---d8754z-;received=192.168.70.10;rport=25702 From: "XLite_PM10";tag=ac93dede To: ;tag=as20e0c424 Call-ID: MGYyNzdhMDAyYzE0NzZlMDFjMmQ1MTcyYzdlM2I1N2Y. CSeq: 2 INVITE Server: Asterisk PBX 1.8.2.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Remote-Party-ID: "PM9 Soft-Phone" ;party=called;privacy=off;screen=no Content-Type: application/sdp Content-Length: 413 v=0 o=root 1789630608 1789630608 IN IP4 192.168.70.51 s=Asterisk PBX 1.8.2.2 c=IN IP4 192.168.70.51 b=CT:8192 t=0 0 m=audio 19650 RTP/AVP 8 98 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 13610 RTP/AVP 34 115 a=rtpmap:34 H263/90000 a=rtpmap:115 h263-1998/90000 a=sendrecv --- Sent RTP P2P packet to 192.168.70.9:63638 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:63638 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:63638 (type 08, len 000160) <--- SIP read from UDP:192.168.70.10:25702 ---> ACK sip:409@192.168.70.51:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.10:25702;branch=z9hG4bK-d8754z-2aa92f969c6b0c26-1---d8754z-;rport Max-Forwards: 70 Contact: To: ;tag=as20e0c424 From: "XLite_PM10";tag=ac93dede Call-ID: MGYyNzdhMDAyYzE0NzZlMDFjMmQ1MTcyYzdlM2I1N2Y. CSeq: 2 ACK User-Agent: X-Lite 4 release 4.0 stamp 58832 Authorization: Digest username="410",realm="phone.lab403.neis.khabarovsk.su",nonce="0e980c3c",uri="sip:409@asterisk.phone.lab403.neis.khabarovsk.su",response="af953fb6c6e590da94f825e0881aaa56",algorithm=MD5 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.70.10:25702 Audio is at 5060 Video is at 192.168.70.51:5060 Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding video codec 0x80000 (h263) to SDP Adding video codec 0x100000 (h263p) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.70.10:25702: INVITE sip:410@192.168.70.10:25702 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK31759080;rport Max-Forwards: 70 From: ;tag=as20e0c424 To: "XLite_PM10";tag=ac93dede Contact: Call-ID: MGYyNzdhMDAyYzE0NzZlMDFjMmQ1MTcyYzdlM2I1N2Y. CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.2.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Remote-Party-ID: "PM9 Soft-Phone" ;party=called;privacy=off;screen=no Content-Type: application/sdp Content-Length: 366 v=0 o=root 1789630608 1789630609 IN IP4 192.168.70.9 s=Asterisk PBX 1.8.2.2 c=IN IP4 192.168.70.9 b=CT:8192 t=0 0 m=audio 63638 RTP/AVP 8 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 49624 RTP/AVP 34 115 a=rtpmap:34 H263/90000 a=rtpmap:115 h263-1998/90000 a=sendrecv --- Sent RTP P2P packet to 192.168.70.9:63638 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:63638 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:63638 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:63638 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:63638 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:63638 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:63638 (type 08, len 000160) Got RTP packet from 192.168.70.10:62960 (type 126, seq 006703, ts 000000, len 000004) [Feb 19 18:05:52] NOTICE[30833]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown RTP codec 126 received from '192.168.70.10:62960' Got RTP packet from 192.168.70.10:62960 (type 126, seq 006703, ts 000000, len 000004) [Feb 19 18:05:52] NOTICE[30833]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown RTP codec 126 received from '192.168.70.10:62960' Got RTP packet from 192.168.70.10:62960 (type 126, seq 006703, ts 000000, len 000004) [Feb 19 18:05:52] NOTICE[30833]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown RTP codec 126 received from '192.168.70.10:62960' Sent RTP P2P packet to 192.168.70.9:63638 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:63638 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:63638 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:63638 (type 08, len 000160) <--- SIP read from UDP:192.168.70.10:25702 ---> ACK sip:409@192.168.70.51:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.10:25702;branch=z9hG4bK-d8754z-2aa92f969c6b0c26-1---d8754z-;rport Max-Forwards: 70 Contact: To: ;tag=as20e0c424 From: "XLite_PM10";tag=ac93dede Call-ID: MGYyNzdhMDAyYzE0NzZlMDFjMmQ1MTcyYzdlM2I1N2Y. CSeq: 2 ACK User-Agent: X-Lite 4 release 4.0 stamp 58832 Authorization: Digest username="410",realm="phone.lab403.neis.khabarovsk.su",nonce="0e980c3c",uri="sip:409@asterisk.phone.lab403.neis.khabarovsk.su",response="af953fb6c6e590da94f825e0881aaa56",algorithm=MD5 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Retransmitting #1 (no NAT) to 192.168.70.10:25702: INVITE sip:410@192.168.70.10:25702 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK31759080;rport Max-Forwards: 70 From: ;tag=as20e0c424 To: "XLite_PM10";tag=ac93dede Contact: Call-ID: MGYyNzdhMDAyYzE0NzZlMDFjMmQ1MTcyYzdlM2I1N2Y. CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.2.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Remote-Party-ID: "PM9 Soft-Phone" ;party=called;privacy=off;screen=no Content-Type: application/sdp Content-Length: 366 v=0 o=root 1789630608 1789630609 IN IP4 192.168.70.9 s=Asterisk PBX 1.8.2.2 c=IN IP4 192.168.70.9 b=CT:8192 t=0 0 m=audio 63638 RTP/AVP 8 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 49624 RTP/AVP 34 115 a=rtpmap:34 H263/90000 a=rtpmap:115 h263-1998/90000 a=sendrecv --- <--- SIP read from UDP:192.168.70.10:25702 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK31759080;rport=5060 Contact: To: "XLite_PM10";tag=ac93dede From: ;tag=as20e0c424 Call-ID: MGYyNzdhMDAyYzE0NzZlMDFjMmQ1MTcyYzdlM2I1N2Y. CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite 4 release 4.0 stamp 58832 Content-Length: 652 v=0 o=- 12942576345423565 2 IN IP4 192.168.70.10 s=CounterPath X-Lite 4.0 c=IN IP4 192.168.70.10 t=0 0 a=ice-ufrag:eae5fa a=ice-pwd:a3d299009f9a92709e8403171a72aa77 m=audio 55966 RTP/AVP 8 3 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.10 55966 typ host a=candidate:1 2 UDP 659134 192.168.70.10 55967 typ host m=video 62960 RTP/AVP 34 115 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=2;CIF=3 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=3;I=1;J=1;T=1 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.10 62960 typ host a=candidate:1 2 UDP 659134 192.168.70.10 62961 typ host <-------------> --- (12 headers 21 lines) --- Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 101 Found audio description format telephone-event for ID 101 Found RTP video format 34 Found RTP video format 115 Found video description format H263 for ID 34 Found video description format H263-1998 for ID 115 Capabilities: us - 0x3c078b (g723|gsm|alaw|lpc10|g729|speex|ilbc|h261|h263|h263p|h264), peer - audio=0xa (gsm|alaw)/video=0x180000 (h263|h263p)/text=0x0 (nothing), combined - 0x18000a (gsm|alaw|h263|h263p) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.70.10:55966 Peer video RTP is at port 192.168.70.10:62960 Peer doesn't provide T.140 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.70.10:25702 Transmitting (no NAT) to 192.168.70.10:25702: ACK sip:410@192.168.70.10:25702 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK27d7d0d4;rport Max-Forwards: 70 From: ;tag=as20e0c424 To: "XLite_PM10";tag=ac93dede Contact: Call-ID: MGYyNzdhMDAyYzE0NzZlMDFjMmQ1MTcyYzdlM2I1N2Y. CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.2.2 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.70.9:38628 Audio is at 5060 Video is at 192.168.70.51:5060 Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding video codec 0x80000 (h263) to SDP Adding video codec 0x100000 (h263p) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.70.9:38628: INVITE sip:409@192.168.70.9:38628;rinstance=51c54eb541dc3435 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK2bbd90f5 Max-Forwards: 70 From: "PM10 Soft-Phone" ;tag=as5c51a7b5 To: ;tag=0f5a5926 Contact: Call-ID: 738ccdde7c12dfbd13a7f0fa3e4a422f@192.168.70.51:5060 CSeq: 104 INVITE User-Agent: Asterisk PBX 1.8.2.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Remote-Party-ID: "PM10 Soft-Phone" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 364 v=0 o=root 692438885 692438887 IN IP4 192.168.70.10 s=Asterisk PBX 1.8.2.2 c=IN IP4 192.168.70.10 b=CT:8192 t=0 0 m=audio 55966 RTP/AVP 8 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 62960 RTP/AVP 34 98 a=rtpmap:34 H263/90000 a=rtpmap:98 h263-1998/90000 a=sendrecv --- <--- SIP read from UDP:192.168.70.9:38628 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK2bbd90f5 To: ;tag=0f5a5926 From: "PM10 Soft-Phone" ;tag=as5c51a7b5 Call-ID: 738ccdde7c12dfbd13a7f0fa3e4a422f@192.168.70.51:5060 CSeq: 104 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from UDP:192.168.70.10:25702 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK31759080;rport=5060 Contact: To: "XLite_PM10";tag=ac93dede From: ;tag=as20e0c424 Call-ID: MGYyNzdhMDAyYzE0NzZlMDFjMmQ1MTcyYzdlM2I1N2Y. CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite 4 release 4.0 stamp 58832 Content-Length: 652 v=0 o=- 12942576345423565 2 IN IP4 192.168.70.10 s=CounterPath X-Lite 4.0 c=IN IP4 192.168.70.10 t=0 0 a=ice-ufrag:eae5fa a=ice-pwd:a3d299009f9a92709e8403171a72aa77 m=audio 55966 RTP/AVP 8 3 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.10 55966 typ host a=candidate:1 2 UDP 659134 192.168.70.10 55967 typ host m=video 62960 RTP/AVP 34 115 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=2;CIF=3 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=3;I=1;J=1;T=1 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.10 62960 typ host a=candidate:1 2 UDP 659134 192.168.70.10 62961 typ host <-------------> --- (12 headers 21 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.70.10:25702 Transmitting (no NAT) to 192.168.70.10:25702: ACK sip:410@192.168.70.10:25702 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK0aef2c2a;rport Max-Forwards: 70 From: ;tag=as20e0c424 To: "XLite_PM10";tag=ac93dede Contact: Call-ID: MGYyNzdhMDAyYzE0NzZlMDFjMmQ1MTcyYzdlM2I1N2Y. CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.2.2 Content-Length: 0 --- <--- SIP read from UDP:192.168.70.9:38628 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK2bbd90f5 Contact: To: ;tag=0f5a5926 From: "PM10 Soft-Phone";tag=as5c51a7b5 Call-ID: 738ccdde7c12dfbd13a7f0fa3e4a422f@192.168.70.51:5060 CSeq: 104 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite 4 release 4.0 stamp 58832 Content-Length: 655 v=0 o=- 12942576351816681 3 IN IP4 192.168.70.9 s=CounterPath X-Lite 4.0 c=IN IP4 192.168.70.9 t=0 0 a=ice-ufrag:9fb215 a=ice-pwd:6e9ec541ff57f81d53879f0807748252 m=audio 63638 RTP/AVP 8 3 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.9 63638 typ host a=candidate:1 2 UDP 659134 192.168.70.9 63639 typ host m=video 49624 RTP/AVP 34 98 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=1;CIF=1;VGA=1 a=rtpmap:98 H263-1998/90000 a=fmtp:98 QCIF=1;CIF=1;VGA=1;I=1;J=1;T=1 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.9 49624 typ host a=candidate:1 2 UDP 659134 192.168.70.9 49625 typ host <-------------> --- (12 headers 21 lines) --- Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 101 Found audio description format telephone-event for ID 101 Found RTP video format 34 Found RTP video format 98 Found video description format H263 for ID 34 Found video description format H263-1998 for ID 98 Capabilities: us - 0x3c078b (g723|gsm|alaw|lpc10|g729|speex|ilbc|h261|h263|h263p|h264), peer - audio=0xa (gsm|alaw)/video=0x180000 (h263|h263p)/text=0x0 (nothing), combined - 0x18000a (gsm|alaw|h263|h263p) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.70.9:63638 Peer video RTP is at port 192.168.70.9:49624 Peer doesn't provide T.140 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.70.9:38628 Transmitting (no NAT) to 192.168.70.9:38628: ACK sip:409@192.168.70.9:38628;rinstance=51c54eb541dc3435 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK1181d4f4 Max-Forwards: 70 From: "PM10 Soft-Phone" ;tag=as5c51a7b5 To: ;tag=0f5a5926 Contact: Call-ID: 738ccdde7c12dfbd13a7f0fa3e4a422f@192.168.70.51:5060 CSeq: 104 ACK User-Agent: Asterisk PBX 1.8.2.2 Content-Length: 0 --- <--- SIP read from UDP:192.168.70.10:25702 ---> BYE sip:409@192.168.70.51:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.10:25702;branch=z9hG4bK-d8754z-dc38f03277394b3d-1---d8754z-;rport Max-Forwards: 70 Contact: To: ;tag=as20e0c424 From: "XLite_PM10";tag=ac93dede Call-ID: MGYyNzdhMDAyYzE0NzZlMDFjMmQ1MTcyYzdlM2I1N2Y. CSeq: 3 BYE User-Agent: X-Lite 4 release 4.0 stamp 58832 Authorization: Digest username="410",realm="phone.lab403.neis.khabarovsk.su",nonce="0e980c3c",uri="sip:409@192.168.70.51:5060",response="474122b2ce6c717fb895f9556fd4519f",algorithm=MD5 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 192.168.70.10:25702 (no NAT) Scheduling destruction of SIP dialog 'MGYyNzdhMDAyYzE0NzZlMDFjMmQ1MTcyYzdlM2I1N2Y.' in 32000 ms (Method: BYE) <--- Transmitting (no NAT) to 192.168.70.10:25702 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.10:25702;branch=z9hG4bK-d8754z-dc38f03277394b3d-1---d8754z-;received=192.168.70.10;rport=25702 From: "XLite_PM10";tag=ac93dede To: ;tag=as20e0c424 Call-ID: MGYyNzdhMDAyYzE0NzZlMDFjMmQ1MTcyYzdlM2I1N2Y. CSeq: 3 BYE Server: Asterisk PBX 1.8.2.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.70.9:38628 Audio is at 5060 Video is at 192.168.70.51:5060 Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding video codec 0x80000 (h263) to SDP Adding video codec 0x100000 (h263p) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.70.9:38628: INVITE sip:409@192.168.70.9:38628;rinstance=51c54eb541dc3435 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK79d653c3 Max-Forwards: 70 From: "PM10 Soft-Phone" ;tag=as5c51a7b5 To: ;tag=0f5a5926 Contact: Call-ID: 738ccdde7c12dfbd13a7f0fa3e4a422f@192.168.70.51:5060 CSeq: 105 INVITE User-Agent: Asterisk PBX 1.8.2.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Remote-Party-ID: "PM10 Soft-Phone" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 364 v=0 o=root 692438885 692438888 IN IP4 192.168.70.51 s=Asterisk PBX 1.8.2.2 c=IN IP4 192.168.70.51 b=CT:8192 t=0 0 m=audio 18966 RTP/AVP 8 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 19764 RTP/AVP 34 98 a=rtpmap:34 H263/90000 a=rtpmap:98 h263-1998/90000 a=sendrecv --- Scheduling destruction of SIP dialog '738ccdde7c12dfbd13a7f0fa3e4a422f@192.168.70.51:5060' in 32000 ms (Method: INVITE) == Spawn extension (phones, 409, 50005) exited non-zero on 'SIP/410-00000000' <--- SIP read from UDP:192.168.70.9:38628 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK79d653c3 Contact: To: ;tag=0f5a5926 From: "PM10 Soft-Phone";tag=as5c51a7b5 Call-ID: 738ccdde7c12dfbd13a7f0fa3e4a422f@192.168.70.51:5060 CSeq: 105 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite 4 release 4.0 stamp 58832 Content-Length: 655 v=0 o=- 12942576351816681 4 IN IP4 192.168.70.9 s=CounterPath X-Lite 4.0 c=IN IP4 192.168.70.9 t=0 0 a=ice-ufrag:9fb215 a=ice-pwd:6e9ec541ff57f81d53879f0807748252 m=audio 63638 RTP/AVP 8 3 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.9 63638 typ host a=candidate:1 2 UDP 659134 192.168.70.9 63639 typ host m=video 49624 RTP/AVP 34 98 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=1;CIF=1;VGA=1 a=rtpmap:98 H263-1998/90000 a=fmtp:98 QCIF=1;CIF=1;VGA=1;I=1;J=1;T=1 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.9 49624 typ host a=candidate:1 2 UDP 659134 192.168.70.9 49625 typ host <-------------> --- (12 headers 21 lines) --- Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 101 Found audio description format telephone-event for ID 101 Found RTP video format 34 Found RTP video format 98 Found video description format H263 for ID 34 Found video description format H263-1998 for ID 98 Capabilities: us - 0x3c078b (g723|gsm|alaw|lpc10|g729|speex|ilbc|h261|h263|h263p|h264), peer - audio=0xa (gsm|alaw)/video=0x180000 (h263|h263p)/text=0x0 (nothing), combined - 0x18000a (gsm|alaw|h263|h263p) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.70.9:63638 Peer video RTP is at port 192.168.70.9:49624 Peer doesn't provide T.140 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.70.9:38628 Transmitting (no NAT) to 192.168.70.9:38628: ACK sip:409@192.168.70.9:38628;rinstance=51c54eb541dc3435 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK10bf696c Max-Forwards: 70 From: "PM10 Soft-Phone" ;tag=as5c51a7b5 To: ;tag=0f5a5926 Contact: Call-ID: 738ccdde7c12dfbd13a7f0fa3e4a422f@192.168.70.51:5060 CSeq: 105 ACK User-Agent: Asterisk PBX 1.8.2.2 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.70.9:38628 Reliably Transmitting (no NAT) to 192.168.70.9:38628: BYE sip:409@192.168.70.9:38628;rinstance=51c54eb541dc3435 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK30b72d27 Max-Forwards: 70 From: "PM10 Soft-Phone" ;tag=as5c51a7b5 To: ;tag=0f5a5926 Call-ID: 738ccdde7c12dfbd13a7f0fa3e4a422f@192.168.70.51:5060 CSeq: 106 BYE User-Agent: Asterisk PBX 1.8.2.2 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- Scheduling destruction of SIP dialog '738ccdde7c12dfbd13a7f0fa3e4a422f@192.168.70.51:5060' in 32000 ms (Method: INVITE) <--- SIP read from UDP:192.168.70.9:38628 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK30b72d27 Contact: To: ;tag=0f5a5926 From: "PM10 Soft-Phone";tag=as5c51a7b5 Call-ID: 738ccdde7c12dfbd13a7f0fa3e4a422f@192.168.70.51:5060 CSeq: 106 BYE User-Agent: X-Lite 4 release 4.0 stamp 58832 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '738ccdde7c12dfbd13a7f0fa3e4a422f@192.168.70.51:5060' Method: INVITE Really destroying SIP dialog 'MGYyNzdhMDAyYzE0NzZlMDFjMmQ1MTcyYzdlM2I1N2Y.' Method: BYE *CLI> *CLI> *CLI> shutdown Waiting for inactivity to perform halt... == Unregistered application 'Monitor' == Unregistered application 'StopMonitor' == Unregistered application 'ChangeMonitor' == Unregistered application 'PauseMonitor' == Unregistered application 'UnpauseMonitor' == Manager unregistered action Monitor == Manager unregistered action StopMonitor == Manager unregistered action ChangeMonitor == Manager unregistered action PauseMonitor == Manager unregistered action UnpauseMonitor == Unregistered custom function FAXOPT == Unregistered application 'SendFAX' == Unregistered application 'ReceiveFAX' == Destroying musiconhold processes == Unregistered application 'MusicOnHold' == Unregistered application 'WaitMusicOnHold' == Unregistered application 'SetMusicOnHold' == Unregistered application 'StartMusicOnHold' == Unregistered application 'StopMusicOnHold' == Unregistered application 'Originate' == Unregistered custom function DEVICE_STATE == Unregistered custom function HINT == Unregistered translator 'lintoulaw' from format slin to ulaw == Unregistered translator 'ulawtolin' from format ulaw to slin == Unregistered translator 'testlawtolin' from format testlaw to slin == Unregistered translator 'lintotestlaw' from format slin to testlaw == Unregistered application 'Echo' == Unregistered format 'jpg' (JPEG (Joint Picture Experts Group)) == Manager unregistered action LocalOptimizeAway == Unregistered channel type 'Local' == Unregistered application 'WaitForRing' == Unregistered format sln16 == Unregistered format siren7 == Unregistered application 'CELGenUserEvent' == Unregistered application 'NoCDR' == Unregistered application 'Record' == Unregistered custom function SHELL == Unregistered application 'ChanIsAvail' == Unregistered application 'DAHDIBarge' == Unregistered application 'ChannelRedirect' == Unregistered application 'Playback' == Unregistered application 'DAHDISendKeypadFacility' == Unregistered application 'DAHDISendCallreroutingFacility' == Unregistered application 'DAHDIAcceptR2Call' == Manager unregistered action DAHDIDialOffhook == Manager unregistered action DAHDIHangup == Manager unregistered action DAHDITransfer == Manager unregistered action DAHDIDNDoff == Manager unregistered action DAHDIDNDon == Manager unregistered action DAHDIShowChannels == Manager unregistered action DAHDIRestart == Unregistered channel type 'DAHDI' -- Unregistered channel -2 == Unregistered custom function VOLUME == Unregistered application 'ReadFile' == Unregistered custom function AST_CONFIG == Unregistered application 'UserEvent' == Unregistered custom function SRVQUERY == Unregistered custom function SRVRESULT == Unregistered application 'ControlPlayback' == Unregistered application 'Page' == Unregistered application 'Exec' == Unregistered application 'TryExec' == Unregistered application 'ExecIf' == Unregistered custom function GLOBAL == Unregistered custom function SHARED == Unregistered custom function ODBC_PRESENCE == Unregistered custom function ODBC_ANTIGF == Unregistered custom function ODBC_SQL == Unregistered custom function SQL_ESC == Unregistered custom function ODBC_FETCH == Unregistered application 'ODBCFinish' == Unregistered application 'Authenticate' == Unregistered application 'ADSIProg' == Unregistered application 'Flash' -- Remove stdexten/a/2, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove stdexten/a/1, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove stdexten/_stdexten-./1, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove stdexten/stdexten-BUSY/3, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove stdexten/stdexten-BUSY/2, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove stdexten/stdexten-BUSY/1, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove stdexten/stdexten-NOANSWER/3, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove stdexten/stdexten-NOANSWER/2, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove stdexten/stdexten-NOANSWER/1, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove stdexten/_X./50006, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove stdexten/_X./50005, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove stdexten/_X./50004, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove stdexten/_X./50003, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove stdexten/_X./50002, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove stdexten/_X./50001, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove stdexten/_X./50000, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove labextens/_4XX/1, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove labextens/_3XX/1, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove labextens/_2XX/1, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove labextens/_1XX/1, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove inbound/t/2, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove inbound/t/1, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove inbound/i/2, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove inbound/i/1, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove inbound/s/6, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove inbound/s/5, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove inbound/s/4, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove inbound/s/3, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove inbound/s/2, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove inbound/s/1, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove outbound-international/_810XXXNXXXXXX/3, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove outbound-international/_810XXXNXXXXXX/2, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove outbound-international/_810XXXNXXXXXX/1, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove outbound-longdistance/_8XXXNXXXXXX/3, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove outbound-longdistance/_8XXXNXXXXXX/2, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove outbound-longdistance/_8XXXNXXXXXX/1, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove outbound-public/_9NXXXXX/3, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove outbound-public/_9NXXXXX/2, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove outbound-public/_9NXXXXX/1, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove outbound-hiik/_7XXX/3, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove outbound-hiik/_7XXX/2, registrar=pbx_config; con=((nil)); con->root=(nil) -- Remove outbound-hiik/_7XXX/1, registrar=pbx_config; con=((nil)); con->root=(nil) == Unregistered RTP engine 'asterisk' == Unregistered application 'System' == Unregistered application 'TrySystem' == Unregistered application 'SetCallerPres' == Unregistered translator 'slin16_to_slin8' from format slin16 to slin == Unregistered translator 'slin8_to_slin16' from format slin to slin16 == Unregistered custom function ICONV == Unregistered format g723sf == Unregistered custom function FRAME_TRACE == Unregistered format iLBC == Unregistered application 'SoftHangup' == Unregistered custom function REALTIME == Unregistered custom function REALTIME_STORE == Unregistered custom function REALTIME_DESTROY == Unregistered custom function REALTIME_FIELD == Unregistered custom function REALTIME_HASH == Unregistered custom function CURL == Unregistered custom function CURLOPT == Unregistered bridge technology simple_bridge == Unregistered custom function SPRINTF == Unregistered custom function DIALPLAN_EXISTS == Unregistered application 'Dictate' == Unregistered custom function GROUP_COUNT == Unregistered custom function GROUP_MATCH_COUNT == Unregistered custom function GROUP_LIST == Unregistered custom function GROUP == Unregistered format siren14 == Unregistered translator 'g722tolin' from format g722 to slin == Unregistered translator 'lintog722' from format slin to g722 == Unregistered translator 'g722tolin16' from format g722 to slin16 == Unregistered translator 'lin16tog722' from format slin16 to g722 == Unregistered application 'ICES' == Unregistered custom function RAND == Unregistered translator 'lintoadpcm' from format slin to adpcm == Unregistered translator 'adpcmtolin' from format adpcm to slin == Unregistered application 'Directory' == Unregistered application 'DISA' == Unregistered application 'NBScat' == Manager unregistered action MeetmeMute == Manager unregistered action MeetmeUnmute == Manager unregistered action MeetmeList == Unregistered application 'MeetMeChannelAdmin' == Unregistered application 'MeetMeAdmin' == Unregistered application 'MeetMeCount' == Unregistered application 'MeetMe' == Unregistered application 'SLAStation' == Unregistered application 'SLATrunk' == Unregistered custom function MEETME_INFO == Unregistered bridge technology multiplexed_bridge == Unregistered custom function DB == Unregistered custom function DB_EXISTS == Unregistered custom function DB_DELETE == Unregistered custom function SYSINFO == Unregistered format sln == Unregistered custom function CHANNEL == Unregistered custom function CHANNELS == Unregistered custom function MASTER_CHANNEL == Unregistered format pcm == Unregistered format alaw == Unregistered format au == Unregistered format g722 == Unregistered application 'SpeechCreate' == Unregistered application 'SpeechLoadGrammar' == Unregistered application 'SpeechUnloadGrammar' == Unregistered application 'SpeechActivateGrammar' == Unregistered application 'SpeechDeactivateGrammar' == Unregistered application 'SpeechStart' == Unregistered application 'SpeechBackground' == Unregistered application 'SpeechDestroy' == Unregistered application 'SpeechProcessingSound' == Unregistered custom function SPEECH == Unregistered custom function SPEECH_SCORE == Unregistered custom function SPEECH_TEXT == Unregistered custom function SPEECH_GRAMMAR == Unregistered custom function SPEECH_ENGINE == Unregistered custom function SPEECH_RESULTS_TYPE == Unregistered format vox == Unregistered application 'SMS' == Unregistered application 'SendText' == Unregistered RTP engine 'multicast' == Unregistered format ogg_vorbis == Unregistered application 'PrivacyManager' == Unregistered application 'ChanSpy' == Unregistered application 'ExtenSpy' == Unregistered application 'DAHDIScan' == Unregistered custom function DIALGROUP == Unregistered application 'SayUnixTime' == Unregistered application 'DateTime' == Unregistered application 'ConfBridge' == Unregistered application 'SendURL' == Unregistered translator 'ulawtoalaw' from format ulaw to alaw == Unregistered translator 'alawtoulaw' from format alaw to ulaw == Unregistered application 'PlayTones' == Unregistered application 'StopPlayTones' == Unregistered application 'Dial' == Unregistered application 'RetryDial' == Unregistered application 'Pickup' == Unregistered application 'PickupChan' == Unregistered custom function EXTENSION_STATE == Unregistered custom function CALLCOMPLETION == Unregistered custom function AES_DECRYPT == Unregistered custom function AES_ENCRYPT == Unregistered channel type 'SIP' == Unregistered custom function SIPCHANINFO == Unregistered custom function SIPPEER == Unregistered custom function SIP_HEADER == Unregistered custom function CHECKSIPDOMAIN == Unregistered application 'SIPDtmfMode' == Unregistered application 'SIPAddHeader' == Unregistered application 'SIPRemoveHeader' == Unregistered RTP glue 'SIP' == Manager unregistered action SIPpeers == Manager unregistered action SIPshowpeer == Manager unregistered action SIPqualifypeer == Manager unregistered action SIPshowregistry == Manager unregistered action SIPnotify Really destroying SIP dialog 'M2E4NzgzYmQxNzQ3ZDFkMmY3YWU0ZTU1OGJkMTFhZTE.' Method: SUBSCRIBE Really destroying SIP dialog 'MmRiODA1MDMyMmZjNjljYWE2YmVjODJkMDZkZWM1YzQ.' Method: SUBSCRIBE == Unregistered application 'Transfer' == Unregistered channel type 'MulticastRTP' == Unregistered translator 'lintolpc10' from format slin to lpc10 == Unregistered translator 'lpc10tolin' from format lpc10 to slin == Unregistered 'cdr_manager' CDR backend == Unregistered custom function CDR == Unregistered application 'StopMixMonitor' == Unregistered application 'MixMonitor' == Manager unregistered action MixMonitorMute == Unregistered custom function MUTEAUDIO == Manager unregistered action MuteAudio == Unregistered custom function ENV == Unregistered custom function STAT == Unregistered custom function FILE == Unregistered custom function FILE_COUNT_LINE == Unregistered custom function FILE_FORMAT == Unregistered custom function VERSION == Unregistered application 'FollowMe' == Unregistered custom function PITCH_SHIFT == Unregistered application 'DBdeltree' == Unregistered application 'DBdel' == Unregistered application 'ReadExten' == Unregistered custom function VALID_EXTEN == Unregistered application 'SendDTMF' == Manager unregistered action PlayDTMF == Unregistered application 'Zapateller' == Unregistered custom function MATH == Unregistered custom function INC == Unregistered custom function DEC -- Security Logging Disabled == Unregistered 'ODBC' CDR backend == Unregistered application 'MP3Player' == Unregistered application 'Read' == Unregistered application 'DAHDIRAS' == Unregistered application 'DumpChan' == Unregistered application 'GetCPEID' == Unregistered custom function VMCOUNT == Unregistered application 'MacroIf' == Unregistered application 'MacroExit' == Unregistered application 'Macro' == Unregistered application 'MacroExclusive' == Unregistered application 'SendImage' == Unregistered custom function BASE64_ENCODE == Unregistered custom function BASE64_DECODE == Unregistered custom function AUDIOHOOK_INHERIT == Unregistered custom function MD5 == Unregistered application 'VoiceMail' == Unregistered application 'VoiceMailMain' == Unregistered application 'MailboxExists' == Unregistered application 'VMAuthenticate' == Unregistered application 'VMSayName' == Unregistered custom function MAILBOX_EXISTS == Manager unregistered action VoicemailUsersList == Unregistered application 'ForkCDR' == Unregistered custom function IFMODULE == Unregistered custom function CALLERPRES == Unregistered custom function CALLERID == Unregistered custom function CONNECTEDLINE == Unregistered custom function REDIRECTING == Unregistered application 'Verbose' == Unregistered application 'Log' == Unregistered format wav49 == Unregistered 'csv' CDR backend == Unregistered application 'ExternalIVR' == Unregistered application 'WaitUntil' == Unregistered format g729 == Unregistered application 'Morsecode' == Unregistered application 'While' == Unregistered application 'EndWhile' == Unregistered application 'ExitWhile' == Unregistered application 'ContinueWhile' == Unregistered format g719 == Unregistered application 'ParkAndAnnounce' == Unregistered channel type 'Bridge' == Unregistered format wav == Unregistered format wav16 == AGI Command 'gosub' unregistered == Unregistered application 'Return' == Unregistered application 'StackPop' == Unregistered application 'GosubIf' == Unregistered application 'Gosub' == Unregistered custom function LOCAL == Unregistered custom function LOCAL_PEEK == Unregistered format h263 == Unregistered custom function URIDECODE == Unregistered custom function URIENCODE == Unregistered custom function CUT == Unregistered custom function SORT == Unregistered format gsm == Unregistered application 'TestClient' == Unregistered application 'TestServer' == Unregistered application 'BackgroundDetect' == Unregistered custom function ISNULL == Unregistered custom function SET == Unregistered custom function EXISTS == Unregistered custom function IF == Unregistered custom function IFTIME == Unregistered custom function IMPORT == Unregistered application 'WaitForSilence' == Unregistered application 'WaitForNoise' == Unregistered translator 'lintoalaw' from format slin to alaw == Unregistered translator 'alawtolin' from format alaw to slin == Unregistered custom function LOCK == Unregistered custom function TRYLOCK == Unregistered custom function UNLOCK == Unregistered custom function ENUMRESULT == Unregistered custom function ENUMQUERY == Unregistered custom function ENUMLOOKUP == Unregistered custom function TXTCIDNAME == Unregistered bridge technology softmix == Unregistered custom function TIMEOUT == Unregistered custom function FIELDQTY == Unregistered custom function FIELDNUM == Unregistered custom function FILTER == Unregistered custom function REPLACE == Unregistered custom function LISTFILTER == Unregistered custom function REGEX == Unregistered custom function ARRAY == Unregistered custom function QUOTE == Unregistered custom function CSV_QUOTE == Unregistered custom function LEN == Unregistered custom function STRFTIME == Unregistered custom function STRPTIME == Unregistered custom function EVAL == Unregistered custom function KEYPADHASH == Unregistered custom function HASHKEYS == Unregistered custom function HASH == Unregistered application 'ClearHash' == Unregistered custom function TOUPPER == Unregistered custom function TOLOWER == Unregistered custom function SHIFT == Unregistered custom function POP == Unregistered custom function PUSH == Unregistered custom function UNSHIFT == Unregistered custom function PASSTHRU == Unregistered format g726-40 == Unregistered format g726-32 == Unregistered format g726-24 == Unregistered format g726-16 == Unregistered custom function SHA1 == Unregistered format h264 == Unregistered translator 'g726tolin' from format g726 to slin == Unregistered translator 'lintog726' from format slin to g726 == Unregistered translator 'g726aal2tolin' from format g726aal2 to slin == Unregistered translator 'lintog726aal2' from format slin to g726aal2 == Unregistered custom function BLACKLIST == Unregistered application 'Milliwatt' == Unregistered translator 'lintogsm' from format slin to gsm == Unregistered translator 'gsmtolin' from format gsm to slin == AGI Command 'answer' unregistered == AGI Command 'asyncagi break' unregistered == AGI Command 'channel status' unregistered == AGI Command 'database del' unregistered == AGI Command 'database deltree' unregistered == AGI Command 'database get' unregistered == AGI Command 'database put' unregistered == AGI Command 'exec' unregistered == AGI Command 'get data' unregistered == AGI Command 'get full variable' unregistered == AGI Command 'get option' unregistered == AGI Command 'get variable' unregistered == AGI Command 'hangup' unregistered == AGI Command 'noop' unregistered == AGI Command 'receive char' unregistered == AGI Command 'receive text' unregistered == AGI Command 'record file' unregistered == AGI Command 'say alpha' unregistered == AGI Command 'say digits' unregistered == AGI Command 'say number' unregistered == AGI Command 'say phonetic' unregistered == AGI Command 'say date' unregistered == AGI Command 'say time' unregistered == AGI Command 'say datetime' unregistered == AGI Command 'send image' unregistered == AGI Command 'send text' unregistered == AGI Command 'set autohangup' unregistered == AGI Command 'set callerid' unregistered == AGI Command 'set context' unregistered == AGI Command 'set extension' unregistered == AGI Command 'set music' unregistered == AGI Command 'set priority' unregistered == AGI Command 'set variable' unregistered == AGI Command 'stream file' unregistered == AGI Command 'control stream file' unregistered == AGI Command 'tdd mode' unregistered == AGI Command 'verbose' unregistered == AGI Command 'wait for digit' unregistered == AGI Command 'speech create' unregistered == AGI Command 'speech set' unregistered == AGI Command 'speech destroy' unregistered == AGI Command 'speech load grammar' unregistered == AGI Command 'speech unload grammar' unregistered == AGI Command 'speech activate grammar' unregistered == AGI Command 'speech deactivate grammar' unregistered == AGI Command 'speech recognize' unregistered == Unregistered application 'EAGI' == Unregistered application 'DeadAGI' == Manager unregistered action AGI == Unregistered application 'AGI' Executing last minute cleanups Asterisk cleanly ending (0). [root@linuxserver boris]#