Asterisk SVN-trunk-r115021, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found  == Parsing '/etc/asterisk/extconfig.conf':  == Found  == Parsing '/etc/asterisk/logger.conf':  == Found  Asterisk Event Logger Started /var/log/asterisk/event_log  Asterisk Dynamic Loader Starting:  == Parsing '/etc/asterisk/modules.conf':  == Found  == Parsing '/etc/asterisk/dnsmgr.conf':  == Found  == Manager registered action Ping  == Manager registered action Events  == Manager registered action Logoff  == Manager registered action Login  == Manager registered action Challenge  == Manager registered action Hangup  == Manager registered action Status  == Manager registered action Setvar  == Manager registered action Getvar  == Manager registered action GetConfig  == Manager registered action GetConfigJSON  == Manager registered action UpdateConfig  == Manager registered action CreateConfig  == Manager registered action ListCategories  == Manager registered action Redirect  == Manager registered action Atxfer  == Manager registered action Originate  == Manager registered action Command  == Manager registered action ExtensionState  == Manager registered action AbsoluteTimeout  == Manager registered action MailboxStatus  == Manager registered action MailboxCount  == Manager registered action ListCommands  == Manager registered action SendText  == Manager registered action UserEvent  == Manager registered action WaitEvent  == Manager registered action CoreSettings  == Manager registered action CoreStatus  == Manager registered action Reload  == Manager registered action CoreShowChannels  == Manager registered action ModuleLoad  == Manager registered action ModuleCheck  == Parsing '/etc/asterisk/manager.conf':  == Found  == Parsing '/etc/asterisk/cdr.conf':  == Found [May 1 18:25:53] NOTICE[7307]: cdr.c:1387 do_reload: CDR simple logging enabled.  == Parsing '/etc/asterisk/rtp.conf':  == Found  == RTP Allocating from port range 5000 -> 32767  == Parsing '/etc/asterisk/udptl.conf':  == Found  == UDPTL allocating from port range 1500 -> 4999  Asterisk PBX Core Initializing  Registering builtin applications:  == Registered custom function 'EXCEPTION'  [Answer]  == Registered application 'Answer'  [BackGround]  == Registered application 'BackGround'  [Busy]  == Registered application 'Busy'  [Congestion]  == Registered application 'Congestion'  [ExecIfTime]  == Registered application 'ExecIfTime'  [Goto]  == Registered application 'Goto'  [GotoIf]  == Registered application 'GotoIf'  [GotoIfTime]  == Registered application 'GotoIfTime'  [ImportVar]  == Registered application 'ImportVar'  [Hangup]  == Registered application 'Hangup'  [Incomplete]  == Registered application 'Incomplete'  [KeepAlive]  == Registered application 'KeepAlive'  [NoOp]  == Registered application 'NoOp'  [Progress]  == Registered application 'Progress'  [RaiseException]  == Registered application 'RaiseException'  [ResetCDR]  == Registered application 'ResetCDR'  [Ringing]  == Registered application 'Ringing'  [SayAlpha]  == Registered application 'SayAlpha'  [SayDigits]  == Registered application 'SayDigits'  [SayNumber]  == Registered application 'SayNumber'  [SayPhonetic]  == Registered application 'SayPhonetic'  [Set]  == Registered application 'Set'  [MSet]  == Registered application 'MSet'  [SetAMAFlags]  == Registered application 'SetAMAFlags'  [Wait]  == Registered application 'Wait'  [WaitExten]  == Registered application 'WaitExten'  == Manager registered action ShowDialPlan  == Registered application 'Bridge'  -- Registered extension context 'parkedcalls'  -- Added extension '' 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  == Parsing '/etc/asterisk/enum.conf':  == Found  Asterisk Dynamic Loader Starting:  == Parsing '/etc/asterisk/modules.conf':  == Found [May 1 18:25:53] NOTICE[7307]: loader.c:873 load_modules: 153 modules will be loaded.  == Parsing '/etc/asterisk/indications.conf':  == Found  -- Registered indication country 'cl'  -- Registered indication country 'tw'  -- Registered indication country 'tw'  -- Registered indication country 'us'  -- Registered indication country 'au'  -- Registered indication country 'fr'  -- Registered indication country 'de'  -- Registered indication country 'nl'  -- Registered indication country 'uk'  -- Registered indication country 'fi'  -- Registered indication country 'no'  -- Registered indication country 'br'  -- Registered indication country 'za'  -- Registered indication country 'it'  -- Registered indication country 'us-o'  -- Registered indication country 'gr'  -- Registered indication country 'ru'  -- Registered indication country 'nz'  -- Setting default indication country to 'us'  == Registered application 'PlayTones'  == Registered application 'StopPlayTones'  res_indications.so => (Region-specific tones)  == Registered custom function 'PP_EACH_USER'  == Registered custom function 'PP_EACH_EXTENSION'  == Parsing '/etc/asterisk/sip.conf':  == Found [May 1 18:25:53] WARNING[7307]: res_phoneprov.c:912 set_config: Unable to load users.cfg  res_phoneprov.so => (HTTP Phone Provisioning)  == 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_ael_share.so => (share-able code for AEL) [May 1 18:25:53] WARNING[7307]: res_config_ldap.c:1393 parse_config: Cannot load configuration res_ldap.conf [May 1 18:25:53] NOTICE[7307]: res_config_ldap.c:1316 load_module: Cannot load LDAP RealTime driver.  res_config_ldap.so => (LDAP realtime interface) [May 1 18:25:53] NOTICE[7307]: config.c:1904 ast_config_engine_register: Registered Config Engine odbc  res_config_odbc loaded.  res_config_odbc.so => (Realtime ODBC configuration)  == Registered custom function 'SMDI_MSG_RETRIEVE'  == Registered custom function 'SMDI_MSG' [May 1 18:25:53] NOTICE[7307]: res_smdi.c:840 smdi_load: Unable to load config smdi.conf: SMDI disabled [May 1 18:25:53] NOTICE[7307]: res_smdi.c:1315 load_module: No SMDI interfaces are available to listen on, not starting SMDI listener.  res_speech.so => (Generic Speech Recognition API)  == AGI Command 'answer' registered  == AGI Command 'channel status' registered  == AGI Command 'database del' registered  == AGI Command 'database deltree' registered  == AGI Command 'database get' registered  == AGI Command 'database put' registered  == AGI Command 'exec' registered  == AGI Command 'get data' registered  == AGI Command 'get full variable' registered  == AGI Command 'get option' registered  == AGI Command 'get variable' registered  == AGI Command 'hangup' registered  == AGI Command 'noop' registered  == AGI Command 'receive char' registered  == AGI Command 'receive text' registered  == AGI Command 'record file' registered  == AGI Command 'say alpha' registered  == AGI Command 'say digits' registered  == AGI Command 'say number' registered  == AGI Command 'say phonetic' registered  == AGI Command 'say date' registered  == AGI Command 'say time' registered  == AGI Command 'say datetime' registered  == AGI Command 'send image' registered  == AGI Command 'send text' registered  == AGI Command 'set autohangup' registered  == AGI Command 'set callerid' registered  == AGI Command 'set context' registered  == AGI Command 'set extension' registered  == AGI Command 'set music' registered  == AGI Command 'set priority' registered  == AGI Command 'set variable' registered  == AGI Command 'stream file' registered  == AGI Command 'control stream file' registered  == AGI Command 'tdd mode' registered  == AGI Command 'verbose' registered  == AGI Command 'wait for digit' registered  == AGI Command 'speech create' registered  == AGI Command 'speech set' registered  == AGI Command 'speech destroy' registered  == AGI Command 'speech load grammar' registered  == AGI Command 'speech unload grammar' registered  == AGI Command 'speech activate grammar' registered  == AGI Command 'speech deactivate grammar' registered  == AGI Command 'speech recognize' registered  == AGI Command 'asyncagi break' registered  == Registered application 'DeadAGI'  == Registered application 'EAGI'  == Manager registered action AGI  == Registered application 'AGI'  res_agi.so => (Asterisk Gateway Interface (AGI))  == Registered custom function 'DIALPLAN_EXISTS'  func_dialplan.so => (Dialplan Context/Extension/Priority Checking Functions)  res_clioriginate.so => (Call origination from the CLI)  == Registered file format h263, extension(s) h263  format_h263.so => (Raw H.263 data)  == Registered custom function 'IAXPEER'  == Registered custom function 'IAXVAR'  == Registered application 'IAX2Provision'  == Manager registered action IAXpeers  == Manager registered action IAXpeerlist  == Manager registered action IAXnetstats  == Parsing '/etc/asterisk/iax.conf':  == Found [May 1 18:25:53] WARNING[7307]: chan_iax2.c:10782 set_config: Invalid tos value, refer to QoS documentation [May 1 18:25:53] WARNING[7307]: chan_iax2.c:10960 set_config: Invalid tos value at line 114, refer to QoS documentation  == Binding IAX2 to default address 0.0.0.0:4569  == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2))  == 10 helper threads started  == IAX Ready and Listening  == Loaded firmware 'iaxy.bin'  == Parsing '/etc/asterisk/iaxprov.conf':  == Found [May 1 18:25:53] WARNING[7307]: iax2-provision.c:326 iax_template_parse: Invalid tos value at line 63, refer to QoS documentation  -- Loaded provisioning template 'default'  chan_iax2.so => (Inter Asterisk eXchange (Ver 2))  == Parsing '/etc/asterisk/codecs.conf':  == Found  == Registered translator 'alawtolin' from format alaw to slin, cost 1  == Registered translator 'lintoalaw' from format slin to alaw, cost 1  codec_alaw.so => (A-law Coder/Decoder)  == Registered custom function 'DEVICE_STATE'  == Registered custom function 'HINT'  func_devstate.so => (Gets or sets a device state in the dialplan)  == Registered application 'Log'  == Registered application 'Verbose'  app_verbose.so => (Send verbose output)  == Registered file format g723sf, extension(s) g723|g723sf  format_g723.so => (G.723.1 Simple Timestamp File Format)  == Registered application 'SetCallerPres'  app_setcallerid.so => (Set CallerID Presentation Application) [May 1 18:25:53] WARNING[7307]: cdr_adaptive_odbc.c:97 load_config: Unable to load cdr_adaptive_odbc.conf. No adaptive ODBC CDRs.  cdr_adaptive_odbc.so => (Adaptive ODBC CDR backend)  == Registered custom function 'ENV'  == Registered custom function 'STAT'  == Registered custom function 'FILE'  func_env.so => (Environment/filesystem dialplan functions)  == Registered application 'ControlPlayback'  app_controlplayback.so => (Control Playback Application)  == Registered custom function 'ICONV'  func_iconv.so => (Charset conversions)  == Parsing '/etc/asterisk/cdr.conf':  == Found  == Registered custom function 'RAND'  func_rand.so => (Random number dialplan function) [May 1 18:25:53] NOTICE[7307]: app_queue.c:5069 reload_queue_rules: No queuerules.conf file found, queues will not follow penalty rules  == Parsing '/etc/asterisk/queues.conf':  == Found  -- Registered extension context 'app_queue_gosub_virtual_context'  -- Added extension 's' priority 1 to app_queue_gosub_virtual_context  == Registered application 'Queue'  == Registered application 'AddQueueMember'  == Registered application 'RemoveQueueMember'  == Registered application 'PauseQueueMember'  == Registered application 'UnpauseQueueMember'  == Registered application 'QueueLog'  == Manager registered action Queues  == Manager registered action QueueStatus  == Manager registered action QueueSummary  == Manager registered action QueueAdd  == Manager registered action QueueRemove  == Manager registered action QueuePause  == Manager registered action QueueLog  == Manager registered action QueuePenalty  == Manager registered action QueueRule  == Registered custom function 'QUEUE_VARIABLES'  == Registered custom function 'QUEUE_MEMBER'  == Registered custom function 'QUEUE_MEMBER_COUNT'  == Registered custom function 'QUEUE_MEMBER_LIST'  == Registered custom function 'QUEUE_WAITING_COUNT'  == Registered custom function 'QUEUE_MEMBER_PENALTY'  app_queue.so => (True Call Queueing)  == Registered application 'ExternalIVR'  app_externalivr.so => (External IVR Interface Application)  == Parsing '/etc/asterisk/cdr_manager.conf':  == Found  cdr_manager.so => (Asterisk Manager Interface CDR Backend)  == Registered application 'SendURL'  app_url.so => (Send URL Applications)  == Parsing '/etc/asterisk/codecs.conf':  == Found  == Registered translator 'g722tolin' from format g722 to slin, cost 1  == Registered translator 'lintog722' from format slin to g722, cost 1 [May 1 18:25:53] WARNING[7307]: translate.c:635 __ast_register_translator: plc_samples 160 format f  == Registered translator 'g722tolin16' from format g722 to slin16, cost 1000  == Registered translator 'lin16tog722' from format slin16 to g722, cost 999  codec_g722.so => (ITU G.722-64kbps G722 Transcoder)  == Registered application 'Milliwatt'  app_milliwatt.so => (Digital Milliwatt (mu-law) Test Application)  res_realtime.so => (Realtime Data Lookup/Rewrite)  == Registered file format sln16, extension(s) sln16  format_sln16.so => (Raw Signed Linear 16KHz Audio support (SLN16))  == Registered file format g726-40, extension(s) g726-40  == Registered file format g726-32, extension(s) g726-32  == Registered file format g726-24, extension(s) g726-24  == Registered file format g726-16, extension(s) g726-16  format_g726.so => (Raw G.726 (16/24/32/40kbps) data)  == Registered custom function 'GROUP_COUNT'  == Registered custom function 'GROUP_MATCH_COUNT'  == Registered custom function 'GROUP_LIST'  == Registered custom function 'GROUP'  func_groupcount.so => (Channel group dialplan functions)  == Registered custom function 'VOLUME'  func_volume.so => (Technology independent volume control)  == Registered application 'ICES'  app_ices.so => (Encode and Stream via icecast and ices)  == Registered custom function 'SYSINFO'  func_sysinfo.so => (System information related functions)  == Registered custom function 'REALTIME'  == Registered custom function 'REALTIME_STORE'  == Registered custom function 'REALTIME_DESTROY'  func_realtime.so => (Read/Write/Store/Destroy values from a RealTime repository)  == Registered application 'Morsecode'  app_morsecode.so => (Morse code)  == Registered custom function 'SHELL'  func_shell.so => (Returns the output of a shell command)  == Parsing '/etc/asterisk/codecs.conf':  == Found  == Registered translator 'g726tolin' from format g726 to slin, cost 1000  == Registered translator 'lintog726' from format slin to g726, cost 1  == Registered translator 'g726aal2tolin' from format g726aal2 to slin, cost 1000  == Registered translator 'lintog726aal2' from format slin to g726aal2, cost 1000  == Registered translator 'g726aal2tog726' from format g726aal2 to g726, cost 1  == Registered translator 'g726tog726aal2' from format g726 to g726aal2, cost 1  codec_g726.so => (ITU G.726-32kbps G726 Transcoder)  == Registered application 'WaitForRing'  app_waitforring.so => (Waits until first ring after time)  == Registered file format g729, extension(s) g729  format_g729.so => (Raw G729 data)  == Registered application 'ChannelRedirect'  app_channelredirect.so => (Redirects a given channel to a dialplan target)  == Registered channel type 'Agent' (Call Agent Proxy Channel)  == Parsing '/etc/asterisk/agents.conf':  == Found  == Registered application 'AgentLogin'  == Registered application 'AgentMonitorOutgoing'  == Manager registered action Agents  == Manager registered action AgentLogoff  == Registered custom function 'AGENT'  chan_agent.so => (Agent Proxy Channel)  == Parsing '/etc/asterisk/codecs.conf':  == Found  == Registered translator 'lpc10tolin' from format lpc10 to slin, cost 1000  == Registered translator 'lintolpc10' from format slin to lpc10, cost 1999  codec_lpc10.so => (LPC10 2.4kbps Coder/Decoder)  res_convert.so => (File format conversion CLI command)  == Parsing '/etc/asterisk/skinny.conf':  == Found  == Skinny listening on 0.0.0.0:2000  == Registered channel type 'Skinny' (Skinny Client Control Protocol (Skinny))  pbx_loopback.so => (Loopback Switch)  == Registered translator 'alawtoulaw' from format alaw to ulaw, cost 1  == Registered translator 'ulawtoalaw' from format ulaw to alaw, cost 1  codec_a_mu.so => (A-law and Mulaw direct Coder/Decoder)  pbx_spool.so => (Outgoing Spool Support)  == Registered application 'TrySystem'  == Registered application 'System'  app_system.so => (Generic System() application)  == Registered custom function 'VERSION'  func_version.so => (Get Asterisk Version/Build Info)  == Registered application 'Exec'  == Registered application 'TryExec'  == Registered application 'ExecIf'  app_exec.so => (Executes dialplan applications)  pbx_realtime.so => (Realtime Switch)  == Parsing '/etc/asterisk/oss.conf':  == Found  == Registered channel type 'Console' (OSS Console Channel Driver)  chan_oss.so => (OSS Console Channel Driver)  == Registered custom function 'MATH'  func_math.so => (Mathematical dialplan function)  == Registered translator 'slin16_to_slin8' from format slin16 to slin, cost 1000  == Registered translator 'slin8_to_slin16' from format slin to slin16, cost 1  codec_resample.so => (SLIN Resampling Codec)  == Registered application 'WaitForSilence'  == Registered application 'WaitForNoise'  app_waitforsilence.so => (Wait For Silence)  == Registered file format wav49, extension(s) WAV|wav49  format_wav_gsm.so => (Microsoft WAV format (Proprietary GSM))  == Parsing '/etc/asterisk/codecs.conf':  == Found  == Registered translator 'adpcmtolin' from format adpcm to slin, cost 1  == Registered translator 'lintoadpcm' from format slin to adpcm, cost 1  codec_adpcm.so => (Adaptive Differential PCM Coder/Decoder)  == Registered file format pcm, extension(s) pcm|ulaw|ul|mu  == Registered file format alaw, extension(s) alaw|al  == Registered file format au, extension(s) au  == Registered file format g722, extension(s) g722  format_pcm.so => (Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G.722 16Khz)  == Registered application 'DumpChan'  app_dumpchan.so => (Dump Info About The Calling Channel) [May 1 18:25:53] ERROR[7307]: chan_unistim.c:5300 reload_config: Unable to load config unistim.conf  == Registered application 'ChanIsAvail'  app_chanisavail.so => (Check channel availability) SIP channel loading...  == Parsing '/etc/asterisk/sip.conf':  == Found  == SIP Listening on 0.0.0.0:5060  == Using SIP CoS mark 4  == Parsing '/etc/asterisk/sip_notify.conf':  == Found [May 1 18:25:53] NOTICE[7307]: chan_sip.c:21970 reload_config: reload_config done...Runtime= 0 sec  == Registered channel type 'SIP' (Session Initiation Protocol (SIP))  == Registered application 'SIPDtmfMode'  == Registered application 'SIPAddHeader'  == Registered custom function 'SIP_HEADER'  == Registered custom function 'SIPPEER'  == Registered custom function 'SIPCHANINFO'  == Registered custom function 'CHECKSIPDOMAIN'  == Manager registered action SIPpeers  == Manager registered action SIPshowpeer  == Manager registered action SIPqualifypeer  == Manager registered action SIPshowregistry  chan_sip.so => (Session Initiation Protocol (SIP))  == Registered application 'GetCPEID'  app_getcpeid.so => (Get ADSI CPE ID)  res_limit.so => (Resource limits)  == Registered custom function 'CUT'  == Registered custom function 'SORT'  func_cut.so => (Cut out information from a string)  == Registered application 'TestClient'  == Registered application 'TestServer'  app_test.so => (Interface Test Application)  == Registered application 'ReadFile'  app_readfile.so => (Stores output of file into a variable)  == Registered custom function 'DB'  == Registered custom function 'DB_EXISTS'  == Registered custom function 'DB_DELETE'  func_db.so => (Database (astdb) related dialplan functions)  == Registered application 'StackPop'  == Registered application 'Return'  == Registered application 'GosubIf'  == Registered application 'Gosub'  == Registered custom function 'LOCAL'  app_stack.so => (Dialplan subroutines (Gosub, Return, etc))  == Registered application 'SendText'  app_sendtext.so => (Send Text Applications)  == Registered application 'BackgroundDetect'  app_talkdetect.so => (Playback with Talk Detection)  == Registered application 'ChanSpy'  == Registered application 'ExtenSpy'  app_chanspy.so => (Listen to the audio of an active channel)  == Registered custom function 'MD5'  func_md5.so => (MD5 digest dialplan functions)  == Registered application 'NBScat'  app_nbscat.so => (Silly NBS Stream Application)  == Registered custom function 'URIDECODE'  == Registered custom function 'URIENCODE'  func_uri.so => (URI encode/decode dialplan functions)  == Registered application 'MixMonitor'  == Registered application 'StopMixMonitor'  app_mixmonitor.so => (Mixed Audio Monitoring Application)  == Registered application 'SoftHangup'  app_softhangup.so => (Hangs up the requested channel)  == Registered application 'MP3Player'  app_mp3.so => (Silly MP3 Application)  == Registered 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 'SMS'  app_sms.so => (SMS/PSTN handler)  == 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 'DBdel'  == Registered application 'DBdeltree'  app_db.so => (Database Access Functions)  == Registered application 'NoCDR'  app_cdr.so => (Tell Asterisk to not maintain a CDR for the current call)  == Registered application 'MacroExit'  == Registered application 'MacroIf'  == Registered application 'MacroExclusive'  == Registered application 'Macro'  app_macro.so => (Extension Macros)  == Registered custom function 'CURL'  func_curl.so => (Load external URL)  == Parsing '/etc/asterisk/festival.conf':  == Found  == Registered application 'Festival'  app_festival.so => (Simple Festival Interface)  == Registered custom function 'AST_CONFIG'  func_config.so => (Asterisk configuration file variable access)  == Registered application 'Dictate'  app_dictate.so => (Virtual Dictation Machine)  == Registered application 'Authenticate'  app_authenticate.so => (Authentication Application)  == Registered application 'ReadExten'  == Registered custom function 'VALID_EXTEN'  app_readexten.so => (Read and evaluate extension validity)  == Registered custom function 'VMCOUNT'  func_vmcount.so => (Indicator for whether a voice mailbox has messages in a given folder.)  == Registered custom function 'CALLERPRES'  == Registered custom function 'CALLERID'  func_callerid.so => (Caller ID related dialplan functions) [May 1 18:25:53] WARNING[7307]: app_followme.c:293 reload_followme: No follow me config file (followme.conf), so no follow me  == Parsing '/etc/asterisk/voicemail.conf':  == Found  == Registered application 'VoiceMail'  == Registered application 'VoiceMailMain'  == Registered application 'MailboxExists'  == Registered application 'VMAuthenticate'  == Registered custom function 'MAILBOX_EXISTS'  == Manager registered action VoicemailUsersList  app_voicemail.so => (Comedian Mail (Voicemail System))  == Registered application 'ForkCDR'  app_forkcdr.so => (Fork The CDR into 2 separate entities)  == Registered custom function 'GLOBAL'  == Registered custom function 'SHARED'  func_global.so => (Variable dialplan functions)  == Registered application 'UserEvent'  app_userevent.so => (Custom User Event Application)  == Registered custom function 'TIMEOUT'  func_timeout.so => (Channel timeout dialplan functions) [May 1 18:25:53] NOTICE[7307]: config.c:1904 ast_config_engine_register: Registered Config Engine curl  res_config_curl loaded.  res_config_curl.so => (Realtime Curl configuration)  == Registered application 'SayUnixTime'  == Registered application 'DateTime'  app_sayunixtime.so => (Say time)  == Registered application 'Record'  app_record.so => (Trivial Record Application)  == Registered file format iLBC, extension(s) ilbc  format_ilbc.so => (Raw iLBC data)  == Parsing '/etc/asterisk/codecs.conf':  == Found  == Registered translator 'ulawtolin' from format ulaw to slin, cost 1  == Registered translator 'lintoulaw' from format slin to ulaw, cost 1  codec_ulaw.so => (mu-Law Coder/Decoder)  == Registered custom function 'CHANNEL'  func_channel.so => (Channel information dialplan function)  == Registered application 'Read'  app_read.so => (Read Variable Application)  == Registered application 'ParkAndAnnounce'  app_parkandannounce.so => (Call Parking and Announce Application)  == Registered custom function 'DIALGROUP'  func_dialgroup.so => (Dialgroup dialplan function)  == Registered file format gsm, extension(s) gsm  format_gsm.so => (Raw GSM data) [May 1 18:25:53] WARNING[7307]: cdr_sqlite3_custom.c:165 load_config: Failed to load configuration file. Module not activated.  == Registered custom function 'CDR'  func_cdr.so => (Call Detail Record (CDR) dialplan function)  == Registered application 'PrivacyManager'  app_privacy.so => (Require phone number to be entered, if no CallerID sent)  == Registered application 'MinivmRecord'  == Registered application 'MinivmGreet'  == Registered application 'MinivmNotify'  == Registered application 'MinivmDelete'  == Registered application 'MinivmAccMess'  == Registered custom function 'MINIVMACCOUNT'  == Registered custom function 'MINIVMCOUNTER' [May 1 18:25:53] WARNING[7307]: app_minivm.c:2399 load_config: Failed to load configuration file. Module activated with default settings.  app_minivm.so => (Mini VoiceMail (A minimal Voicemail e-mail System))  == Registered channel type 'Local' (Local Proxy Channel Driver)  chan_local.so => (Local Proxy Channel (Note: used internally by other modules))  == Registered application 'Playback'  app_playback.so => (Sound File Playback Application)  == Manager registered action PlayDTMF  == Registered application 'SendDTMF'  app_senddtmf.so => (Send DTMF digits Application)  == Registered application 'WaitUntil'  app_waituntil.so => (Wait until specified time)  == Registered application 'Zapateller'  app_zapateller.so => (Block Telemarketers with Special Information Tone)  == Parsing '/etc/asterisk/h323.conf':  == Found  == Creating H.323 Endpoint  == Setting default context to default  == Registered channel type 'H323' (The NuFone Network's Open H.323 Channel Driver)  == H.323 listener started  chan_h323.so => (The NuFone Network's OpenH323 Channel Driver)  == Registered custom function 'SHA1'  func_sha1.so => (SHA-1 computation dialplan function)  == Registered custom function 'FIELDQTY'  == Registered custom function 'FILTER'  == Registered custom function 'REGEX'  == Registered custom function 'ARRAY'  == Registered custom function 'QUOTE'  == Registered custom function 'LEN'  == Registered custom function 'STRFTIME'  == Registered custom function 'STRPTIME'  == Registered custom function 'EVAL'  == Registered custom function 'KEYPADHASH'  == Registered custom function 'SPRINTF'  == Registered custom function 'HASHKEYS'  == Registered custom function 'HASH'  == Registered application 'ClearHash'  == Registered custom function 'TOUPPER'  == Registered custom function 'TOLOWER'  func_strings.so => (String handling dialplan functions)  == Registered application 'DISA'  app_disa.so => (DISA (Direct Inward System Access) Application)  == Registered custom function 'BASE64_ENCODE'  == Registered custom function 'BASE64_DECODE'  func_base64.so => (base64 encode/decode dialplan functions)  == Registered custom function 'BLACKLIST'  func_blacklist.so => (Look up Caller*ID name/number from blacklist database)  == Parsing '/etc/asterisk/alarmreceiver.conf':  == Found  == Registered application 'AlarmReceiver'  app_alarmreceiver.so => (Alarm Receiver for Asterisk)  == Registered application 'Transfer'  app_transfer.so => (Transfers a caller to another extension)  == Registered application 'SpeechCreate'  == Registered application 'SpeechLoadGrammar'  == Registered application 'SpeechUnloadGrammar'  == Registered application 'SpeechActivateGrammar'  == Registered application 'SpeechDeactivateGrammar'  == Registered application 'SpeechStart'  == Registered application 'SpeechBackground'  == Registered application 'SpeechDestroy'  == Registered application 'SpeechProcessingSound'  == Registered custom function 'SPEECH'  == Registered custom function 'SPEECH_SCORE'  == Registered custom function 'SPEECH_TEXT'  == Registered custom function 'SPEECH_GRAMMAR'  == Registered custom function 'SPEECH_ENGINE'  == Registered custom function 'SPEECH_RESULTS_TYPE'  app_speech_utils.so => (Dialplan Speech Applications)  == Registered custom function 'IFMODULE'  func_module.so => (Checks if Asterisk module is loaded in memory)  == Registered file format h264, extension(s) h264  format_h264.so => (Raw H.264 data)  == Parsing '/etc/asterisk/phone.conf':  == Found  == Registered channel type 'Phone' (Standard Linux Telephony API Driver)  chan_phone.so => (Linux Telephony API Support)  == Registered custom function 'LOCK'  == Registered custom function 'TRYLOCK'  == Registered custom function 'UNLOCK'  func_lock.so => (Dialplan mutexes)  == Registered application 'Directory'  app_directory.so => (Extension Directory)  == Registered file format vox, extension(s) vox  format_vox.so => (Dialogic VOX (ADPCM) File Format)  == Registered file format wav, extension(s) wav  format_wav.so => (Microsoft WAV format (8000Hz Signed Linear))  == Parsing '/etc/asterisk/codecs.conf':  == Found  == Registered translator 'gsmtolin' from format gsm to slin, cost 1000  == Registered translator 'lintogsm' from format slin to gsm, cost 1000  codec_gsm.so => (GSM Coder/Decoder)  == Registered application 'ADSIProg'  app_adsiprog.so => (Asterisk ADSI Programming Application)  == Registered custom function 'EXTENSION_STATE'  func_extstate.so => (Gets an extension's state in the dialplan)  == Parsing '/etc/asterisk/extensions.conf':  == Found  -- Registered extension context 'default'  -- Added extension '_X.' priority 1 to default  -- Added extension '_X.' priority 2 to default  -- Added extension '_X.' priority 3 to default  -- Added extension '_X.' priority 4 to default  -- Added extension '_X.' priority 5 to default  -- Added extension '_X.' priority 6 to default  -- Added extension '_X.' priority 7 to default  -- Added extension '_X.' priority 8 to default  -- Added extension '_X.' priority 9 to default  -- Added extension '_X.' priority 10 to default  -- Added extension '_X.' priority 100 to default  -- Added extension '_X.' priority 300 to default  -- Added extension '_X.' priority 301 to default  -- Added extension '_X.' priority 302 to default  -- Registered extension context 'default204'  -- Added extension '_X.' priority 1 to default204  -- Added extension '_X.' priority 2 to default204  -- Added extension '_X.' priority 3 to default204  -- Added extension '_X.' priority 4 to default204  -- Added extension '_X.' priority 5 to default204  -- Added extension '_X.' priority 6 to default204  -- Added extension '_X.' priority 7 to default204  -- Added extension '_X.' priority 8 to default204  -- Added extension '_X.' priority 9 to default204  -- Added extension '_X.' priority 10 to default204  -- Added extension '_X.' priority 11 to default204  -- Added extension '_X.' priority 12 to default204  -- Added extension 'closecall' priority 1 to default204  -- Added extension 'closecall' priority 2 to default204  -- Registered extension context 'sendmail'  -- Added extension '_X.' priority 1 to sendmail  -- Added extension '_X.' priority 2 to sendmail  -- Added extension 'h' priority 1 to sendmail  -- Added extension 'h' priority 2 to sendmail  -- Registered extension context 'process'  -- Added extension '_X.' priority 1 to process  -- Added extension '_X.' priority 2 to process  -- Added extension '_X.' priority 3 to process  -- Added extension '_X.' priority 4 to process  -- Added extension '_X.' priority 5 to process  -- Added extension '_X.' priority 6 to process  -- Added extension '_X.' priority 7 to process  -- Added extension '_X.' priority 8 to process  -- Added extension '_X.' priority 9 to process  -- Added extension '_X.' priority 100 to process  -- Added extension '_X.' priority 300 to process  -- Added extension '_X.' priority 301 to process  -- Added extension '_X.' priority 302 to process  -- Registered extension context 'app_queue_gosub_virtual_context'  -- Added extension 's' priority 1 to app_queue_gosub_virtual_context  -- Registered extension context 'parkedcalls'  -- Added extension '700' priority 1 to parkedcalls  -- Added extension '' priority 1 to parkedcalls  -- Time to scan old dialplan and merge leftovers back into the new: 0.000049 sec  -- Time to restore hints and swap in new dialplan: 0.000001 sec  -- Time to delete the old dialplan: 0.000021 sec  -- Total time merge_contexts_delete: 0.000071 sec  pbx_config.so => (Text Extension Configuration)  == Registered format 'jpg' (JPEG (Joint Picture Experts Group))  format_jpeg.so => (JPEG (Joint Picture Experts Group) Image Format) [May 1 18:25:53] WARNING[7307]: cdr_custom.c:96 load_config: Failed to load configuration file. Module not activated. [May 1 18:25:53] NOTICE[7307]: pbx_ael.c:115 pbx_load_module: Starting AEL load process. [May 1 18:25:53] NOTICE[7307]: pbx_ael.c:123 pbx_load_module: File /etc/asterisk/extensions.ael not found; AEL declining load  == Registered application 'Pickup'  == Registered application 'PickupChan'  app_directed_pickup.so => (Directed Call Pickup Application)  == Parsing '/etc/asterisk/adsi.conf':  == Found  res_adsi.so => (ADSI Resource)  == Registered application 'SendImage'  app_image.so => (Image Transmission Application)  -- Loaded PUBLIC key 'iaxtel'  -- Loaded PUBLIC key 'freeworlddialup'  res_crypto.so => (Cryptographic Digital Signatures)  == Registered file format sln, extension(s) sln|raw  format_sln.so => (Raw Signed Linear Audio support (SLN))  -- Registered extension context 'app_dial_gosub_virtual_context'  -- Added extension 's' priority 1 to app_dial_gosub_virtual_context  == Registered application 'Dial'  == Registered application 'RetryDial'  app_dial.so => (Dialing Application)  == Registered application 'While'  == Registered application 'EndWhile'  == Registered application 'ExitWhile'  == Registered application 'ContinueWhile'  app_while.so => (While Loops and Conditional Execution)  == Registered application 'Echo'  app_echo.so => (Simple Echo Application)  == Parsing '/etc/asterisk/mgcp.conf':  == Found  == MGCP Listening on 0.0.0.0:2727  == Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP))  chan_mgcp.so => (Media Gateway Control Protocol (MGCP)) Asterisk Ready. *CLI> [May 1 18:25:53] NOTICE[7307]: chan_sip.c:16115 handle_response_peerpoke: Peer 'sipserver204' is now Reachable. (1ms / 8000ms) [May 1 18:25:53] NOTICE[7307]: chan_sip.c:16115 handle_response_peerpoke: Peer 'sipserver' is now Reachable. (9ms / 8000ms) [May 1 18:25:53] NOTICE[7307]: chan_sip.c:16115 handle_response_peerpoke: Peer '7864335989' is now Reachable. (84ms / 8000ms) sip set debug on SIP Debugging enabled *CLI> <--- SIP read from UDP://72.187.243.97:45990 ---> REGISTER sip:66.28.190.219 SIP/2.0 Via: SIP/2.0/UDP 72.187.243.97:45990;branch=z9hG4bK5174D5373A1F4A3DB97A69C0327128EE From: 7864335989 ;tag=1867072463 To: 7864335989 Contact: "7864335989" Call-ID: FF611B51FF2F4A3B9715B926FB8E7F7A@66.28.190.219 CSeq: 65631 REGISTER Expires: 60 Max-Forwards: 70 User-Agent: X-PRO release 1105x Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 72.187.243.97 : 45990 (NAT)  > Saved useragent "X-PRO release 1105x" for peer 7864335989 <--- Transmitting (NAT) to 72.187.243.97:45990 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 72.187.243.97:45990;branch=z9hG4bK5174D5373A1F4A3DB97A69C0327128EE;received=72.187.243.97 From: 7864335989 ;tag=1867072463 To: 7864335989 ;tag=as5a3e32f5 Call-ID: FF611B51FF2F4A3B9715B926FB8E7F7A@66.28.190.219 CSeq: 65631 REGISTER Server: Cisco 3845 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Expires: 60 Contact: ;expires=60 Date: Thu, 01 May 2008 22:25:56 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'FF611B51FF2F4A3B9715B926FB8E7F7A@66.28.190.219' in 32000 ms (Method: REGISTER) <--- SIP read from UDP://72.187.243.97:45990 ---> INVITE sip:14166289921@66.28.190.219 SIP/2.0 Via: SIP/2.0/UDP 72.187.243.97:45990;branch=z9hG4bK125375140A304CDD9F0C3992E360FC1C From: 7864335989 ;tag=3763291073 To: Contact: Call-ID: 29FAD0E6-5458-4A60-AE64-7617D04CF35E@192.168.1.111 CSeq: 50124 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-PRO release 1105x Content-Length: 217 v=0 o=7864335989 144494472 144494557 IN IP4 72.187.243.97 s=X-PRO c=IN IP4 72.187.243.97 t=0 0 m=audio 45992 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> --- (11 headers 10 lines) ---  == Using SIP RTP CoS mark 5  == Using UDPTL CoS mark 5 Sending to 72.187.243.97 : 45990 (NAT) Using INVITE request as basis request - 29FAD0E6-5458-4A60-AE64-7617D04CF35E@192.168.1.111 No user '7864335989' in SIP users list Found peer '7864335989' for '7864335989' from 72.187.243.97:45990 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 72.187.243.97:45992 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 72.187.243.97:45992 Looking for 14166289921 in default (domain 66.28.190.219) list_route: hop: <--- Transmitting (NAT) to 72.187.243.97:45990 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 72.187.243.97:45990;branch=z9hG4bK125375140A304CDD9F0C3992E360FC1C;received=72.187.243.97 From: 7864335989 ;tag=3763291073 To: Call-ID: 29FAD0E6-5458-4A60-AE64-7617D04CF35E@192.168.1.111 CSeq: 50124 INVITE Server: Cisco 3845 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------>  -- Executing [14166289921@default:1] NoOp("SIP/7864335989-11c76218", "SIP") in new stack  -- Executing [14166289921@default:2] GotoIf("SIP/7864335989-11c76218", "0?default,14166289921,300") in new stack  -- Executing [14166289921@default:3] Set("SIP/7864335989-11c76218", "SIPIP=72.187.243.97") in new stack  -- Executing [14166289921@default:4] Verbose("SIP/7864335989-11c76218", "From= 72.187.243.97 Number= 14166289921") in new stack From= 72.187.243.97 Number= 14166289921  -- Executing [14166289921@default:5] GotoIf("SIP/7864335989-11c76218", "1?default,14166289921,entrada") in new stack  -- Goto (default,14166289921,7)  -- Executing [14166289921@default:7] Set("SIP/7864335989-11c76218", "CALLERID(num)=7864335989") in new stack  -- Executing [14166289921@default:8] NoOp("SIP/7864335989-11c76218", "Callerid: 7864335989") in new stack  -- Executing [14166289921@default:9] Dial("SIP/7864335989-11c76218", "SIP/14166289921@67.110.179.253") in new stack  == Using SIP RTP CoS mark 5  == Using UDPTL CoS mark 5 Audio is at 66.28.190.219 port 10514 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 67.110.179.253:5060: INVITE sip:14166289921@67.110.179.253 SIP/2.0 Via: SIP/2.0/UDP 66.28.190.219:5060;branch=z9hG4bK42b717b6 Max-Forwards: 70 From: "7864335989" ;tag=as191d4943 To: Contact: Call-ID: 04648d943bffb43937163c9e6a3fd210@minixel CSeq: 102 INVITE User-Agent: Cisco 3845 Date: Thu, 01 May 2008 22:26:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 291 v=0 o=root 866673182 866673182 IN IP4 66.28.190.219 s=Cisco-SIPGateway/IOS-12.x c=IN IP4 66.28.190.219 t=0 0 m=audio 10514 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv ---  -- Called 14166289921@67.110.179.253 <--- SIP read from UDP://67.110.179.253:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 66.28.190.219:5060;branch=z9hG4bK42b717b6 From: "7864335989" ;tag=as191d4943 To: Date: Thu, 01 May 2008 22:26:00 GMT Call-ID: 04648d943bffb43937163c9e6a3fd210@minixel Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow-Events: telephone-event Content-Length: 0 <-------------> --- (10 headers 0 lines) --- rtp set debug on RTP Debugging Enabled *CLI> Really destroying SIP dialog 'FF611B51FF2F4A3B9715B926FB8E7F7A@66.28.190.219' Method: REGISTER *CLI> *CLI> *CLI> *CLI> <--- SIP read from UDP://72.187.243.97:45990 ---> CANCEL sip:14166289921@66.28.190.219 SIP/2.0 Via: SIP/2.0/UDP 72.187.243.97:45990;branch=z9hG4bK125375140A304CDD9F0C3992E360FC1C From: 7864335989 ;tag=3763291073 To: Contact: Call-ID: 29FAD0E6-5458-4A60-AE64-7617D04CF35E@192.168.1.111 CSeq: 50124 CANCEL Max-Forwards: 70 User-Agent: X-PRO release 1105x Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 72.187.243.97 : 45990 (NAT) <--- Reliably Transmitting (NAT) to 72.187.243.97:45990 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 72.187.243.97:45990;branch=z9hG4bK125375140A304CDD9F0C3992E360FC1C;received=72.187.243.97 From: 7864335989 ;tag=3763291073 To: ;tag=as57c6a23d Call-ID: 29FAD0E6-5458-4A60-AE64-7617D04CF35E@192.168.1.111 CSeq: 50124 INVITE Server: Cisco 3845 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 <------------> <--- Transmitting (NAT) to 72.187.243.97:45990 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 72.187.243.97:45990;branch=z9hG4bK125375140A304CDD9F0C3992E360FC1C;received=72.187.243.97 From: 7864335989 ;tag=3763291073 To: ;tag=as57c6a23d Call-ID: 29FAD0E6-5458-4A60-AE64-7617D04CF35E@192.168.1.111 CSeq: 50124 CANCEL Server: Cisco 3845 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> Scheduling destruction of SIP dialog '04648d943bffb43937163c9e6a3fd210@minixel' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 67.110.179.253:5060: CANCEL sip:14166289921@67.110.179.253 SIP/2.0 Via: SIP/2.0/UDP 66.28.190.219:5060;branch=z9hG4bK42b717b6 Max-Forwards: 70 From: "7864335989" ;tag=as191d4943 To: Call-ID: 04648d943bffb43937163c9e6a3fd210@minixel CSeq: 102 CANCEL User-Agent: Cisco 3845 Content-Length: 0 --- Scheduling destruction of SIP dialog '04648d943bffb43937163c9e6a3fd210@minixel' in 32000 ms (Method: INVITE)  == Spawn extension (default, 14166289921, 9) exited non-zero on 'SIP/7864335989-11c76218' <--- SIP read from UDP://67.110.179.253:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 66.28.190.219:5060;branch=z9hG4bK42b717b6 From: "7864335989" ;tag=as191d4943 To: Date: Thu, 01 May 2008 22:26:33 GMT Call-ID: 04648d943bffb43937163c9e6a3fd210@minixel CSeq: 102 CANCEL Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP://67.110.179.253:5060 ---> SIP/2.0 487 Request Cancelled Via: SIP/2.0/UDP 66.28.190.219:5060;branch=z9hG4bK42b717b6 From: "7864335989" ;tag=as191d4943 To: ;tag=422F84C-1A00 Date: Thu, 01 May 2008 22:26:33 GMT Call-ID: 04648d943bffb43937163c9e6a3fd210@minixel Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow-Events: telephone-event Reason: Q.850;cause=16 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Transmitting (NAT) to 67.110.179.253:5060: ACK sip:14166289921@67.110.179.253 SIP/2.0 Via: SIP/2.0/UDP 66.28.190.219:5060;branch=z9hG4bK42b717b6 Max-Forwards: 70 From: "7864335989" ;tag=as191d4943 To: ;tag=422F84C-1A00 Contact: Call-ID: 04648d943bffb43937163c9e6a3fd210@minixel CSeq: 102 ACK User-Agent: Cisco 3845 Content-Length: 0 --- Really destroying SIP dialog '04648d943bffb43937163c9e6a3fd210@minixel' Method: INVITE <--- SIP read from UDP://72.187.243.97:45990 ---> ACK sip:14166289921@66.28.190.219 SIP/2.0 Via: SIP/2.0/UDP 72.187.243.97:45990;branch=z9hG4bK125375140A304CDD9F0C3992E360FC1C From: 7864335989 ;tag=3763291073 To: ;tag=as57c6a23d Contact: Call-ID: 29FAD0E6-5458-4A60-AE64-7617D04CF35E@192.168.1.111 CSeq: 50124 ACK Max-Forwards: 70 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '29FAD0E6-5458-4A60-AE64-7617D04CF35E@192.168.1.111' Method: ACK <--- SIP read from UDP://72.187.243.97:45990 ---> ACK sip:14166289921@66.28.190.219 SIP/2.0 Via: SIP/2.0/UDP 72.187.243.97:45990;branch=z9hG4bK125375140A304CDD9F0C3992E360FC1C From: 7864335989 ;tag=3763291073 To: ;tag=as57c6a23d Contact: Call-ID: 29FAD0E6-5458-4A60-AE64-7617D04CF35E@192.168.1.111 CSeq: 50124 ACK Max-Forwards: 70 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP://72.187.243.97:45990 ---> INVITE sip:14166289921@66.28.190.219 SIP/2.0 Via: SIP/2.0/UDP 72.187.243.97:45990;branch=z9hG4bK8EC48F6FEFB0407B92E8155BDDB76937 From: 7864335989 ;tag=1888197430 To: Contact: Call-ID: 99A84C56-D09C-498F-B98D-36F1B01D591A@192.168.1.111 CSeq: 5598 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-PRO release 1105x Content-Length: 217 v=0 o=7864335989 144529377 144529471 IN IP4 72.187.243.97 s=X-PRO c=IN IP4 72.187.243.97 t=0 0 m=audio 45992 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> --- (11 headers 10 lines) ---  == Using SIP RTP CoS mark 5  == Using UDPTL CoS mark 5 Sending to 72.187.243.97 : 45990 (NAT) Using INVITE request as basis request - 99A84C56-D09C-498F-B98D-36F1B01D591A@192.168.1.111 No user '7864335989' in SIP users list Found peer '7864335989' for '7864335989' from 72.187.243.97:45990 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 72.187.243.97:45992 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 72.187.243.97:45992 Looking for 14166289921 in default (domain 66.28.190.219) list_route: hop: <--- Transmitting (NAT) to 72.187.243.97:45990 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 72.187.243.97:45990;branch=z9hG4bK8EC48F6FEFB0407B92E8155BDDB76937;received=72.187.243.97 From: 7864335989 ;tag=1888197430 To: Call-ID: 99A84C56-D09C-498F-B98D-36F1B01D591A@192.168.1.111 CSeq: 5598 INVITE Server: Cisco 3845 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------>  -- Executing [14166289921@default:1] NoOp("SIP/7864335989-11c76218", "SIP") in new stack  -- Executing [14166289921@default:2] GotoIf("SIP/7864335989-11c76218", "0?default,14166289921,300") in new stack  -- Executing [14166289921@default:3] Set("SIP/7864335989-11c76218", "SIPIP=72.187.243.97") in new stack  -- Executing [14166289921@default:4] Verbose("SIP/7864335989-11c76218", "From= 72.187.243.97 Number= 14166289921") in new stack From= 72.187.243.97 Number= 14166289921  -- Executing [14166289921@default:5] GotoIf("SIP/7864335989-11c76218", "1?default,14166289921,entrada") in new stack  -- Goto (default,14166289921,7)  -- Executing [14166289921@default:7] Set("SIP/7864335989-11c76218", "CALLERID(num)=7864335989") in new stack  -- Executing [14166289921@default:8] NoOp("SIP/7864335989-11c76218", "Callerid: 7864335989") in new stack  -- Executing [14166289921@default:9] Dial("SIP/7864335989-11c76218", "SIP/14166289921@67.110.179.253") in new stack  == Using SIP RTP CoS mark 5  == Using UDPTL CoS mark 5 Audio is at 66.28.190.219 port 28352 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 67.110.179.253:5060: INVITE sip:14166289921@67.110.179.253 SIP/2.0 Via: SIP/2.0/UDP 66.28.190.219:5060;branch=z9hG4bK71b38b31 Max-Forwards: 70 From: "7864335989" ;tag=as0da18e6c To: Contact: Call-ID: 3f9f6b31343bbd6b1ef7ab55068e23d5@minixel CSeq: 102 INVITE User-Agent: Cisco 3845 Date: Thu, 01 May 2008 22:26:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 291 v=0 o=root 225492555 225492555 IN IP4 66.28.190.219 s=Cisco-SIPGateway/IOS-12.x c=IN IP4 66.28.190.219 t=0 0 m=audio 28352 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv ---  -- Called 14166289921@67.110.179.253 <--- SIP read from UDP://67.110.179.253:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 66.28.190.219:5060;branch=z9hG4bK71b38b31 From: "7864335989" ;tag=as0da18e6c To: Date: Thu, 01 May 2008 22:26:35 GMT Call-ID: 3f9f6b31343bbd6b1ef7ab55068e23d5@minixel Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow-Events: telephone-event Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Got RTP packet from 66.234.181.135:58680 (type 18, seq 000001, ts 3180022096, len 000020) Got RTP packet from 66.234.181.135:58680 (type 18, seq 000002, ts 3180022256, len 000020) Got RTP packet from 66.234.181.135:58680 (type 18, seq 000001, ts 3180022576, len 000020) Got RTP packet from 66.234.181.135:58680 (type 18, seq 000002, ts 3180022736, len 000020) Got RTP packet from 66.234.181.135:58680 (type 18, seq 000003, ts 3180022896, len 000020) Got RTP packet from 66.234.181.135:58680 (type 18, seq 000004, ts 3180023056, len 000020) Got RTP packet from 66.234.181.135:58680 (type 18, seq 000005, ts 3180023216, len 000020) <--- SIP read from UDP://67.110.179.253:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 66.28.190.219:5060;branch=z9hG4bK71b38b31 From: "7864335989" ;tag=as0da18e6c To: ;tag=4231628-53 Date: Thu, 01 May 2008 22:26:35 GMT Call-ID: 3f9f6b31343bbd6b1ef7ab55068e23d5@minixel Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Remote-Party-ID: ;party=called;screen=no;privacy=off Contact: Supported: replaces Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 303 v=0 o=CiscoSystemsSIP-GW-UserAgent 7528 3553 IN IP4 67.110.179.253 s=SIP Call c=IN IP4 66.234.181.135 t=0 0 m=audio 58680 RTP/AVP 18 101 c=IN IP4 66.234.181.135 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=silenceSupp:off - - - - <-------------> --- (16 headers 13 lines) --- Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 66.234.181.135:58680 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 66.234.181.135:58680 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 67.110.179.253, port 5060 Transmitting (NAT) to 67.110.179.253:5060: ACK sip:14166289921@67.110.179.253:5060 SIP/2.0 Via: SIP/2.0/UDP 66.28.190.219:5060;branch=z9hG4bK4d2630f2 Max-Forwards: 70 From: "7864335989" ;tag=as0da18e6c To: ;tag=4231628-53 Contact: Call-ID: 3f9f6b31343bbd6b1ef7ab55068e23d5@minixel CSeq: 102 ACK User-Agent: Cisco 3845 Content-Length: 0 ---  -- SIP/67.110.179.253-b005a368 answered SIP/7864335989-11c76218 Audio is at 66.28.190.219 port 20144 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 72.187.243.97:45990 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 72.187.243.97:45990;branch=z9hG4bK8EC48F6FEFB0407B92E8155BDDB76937;received=72.187.243.97 From: 7864335989 ;tag=1888197430 To: ;tag=as5822860e Call-ID: 99A84C56-D09C-498F-B98D-36F1B01D591A@192.168.1.111 CSeq: 5598 INVITE Server: Cisco 3845 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 289 v=0 o=root 82575450 82575450 IN IP4 66.28.190.219 s=Cisco-SIPGateway/IOS-12.x c=IN IP4 66.28.190.219 t=0 0 m=audio 20144 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> <--- SIP read from UDP://72.187.243.97:45990 ---> ACK sip:14166289921@66.28.190.219 SIP/2.0 Via: SIP/2.0/UDP 72.187.243.97:45990;branch=z9hG4bKE9474F048701464F8233D077EF67370B From: 7864335989 ;tag=1888197430 To: ;tag=as5822860e Contact: Call-ID: 99A84C56-D09C-498F-B98D-36F1B01D591A@192.168.1.111 CSeq: 5598 ACK Max-Forwards: 70 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Got RTP packet from 72.187.243.97:45992 (type 18, seq 000001, ts 045280, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000002, ts 045360, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000003, ts 045440, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000004, ts 045520, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000005, ts 045600, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000006, ts 045680, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000007, ts 045760, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000008, ts 045840, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000009, ts 045920, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000010, ts 046000, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000011, ts 046080, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000012, ts 046160, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000013, ts 046240, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000014, ts 046320, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000015, ts 046400, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000016, ts 046480, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000017, ts 046560, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000018, ts 046640, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000019, ts 046720, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000020, ts 046800, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000021, ts 046880, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000022, ts 046960, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000023, ts 047040, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000024, ts 047120, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000025, ts 047200, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000026, ts 047280, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000027, ts 047360, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000028, ts 047440, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000029, ts 047520, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000030, ts 047600, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000031, ts 047680, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000032, ts 047760, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000033, ts 047840, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000034, ts 047920, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000035, ts 048000, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000036, ts 048080, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000037, ts 048160, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000038, ts 048240, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000039, ts 048320, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000040, ts 048400, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000041, ts 048480, len 000010) Got RTP packet from 72.187.243.97:45992 (type 18, seq 000042, ts 048560, len 000010)  -- Packet2Packet bridging SIP/7864335989-11c76218 and SIP/67.110.179.253-b005a368 Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 66.234.181.135:58680 (type 18, len 000010) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) Sent RTP P2P packet to 72.187.243.97:45992 (type 18, len 000020) rtp set debug off RTP Debugging Disabled *CLI> rtp set debug off RTP Debugging Disabled *CLI> rtp set debug offnReliably Transmitting (no NAT) to 66.28.147.204:5060: OPTIONS sip:66.28.147.204 SIP/2.0 Via: SIP/2.0/UDP 66.28.190.219:5060;branch=z9hG4bK63d8320d;rport Max-Forwards: 70 From: "Cisco 3845" ;tag=as3c1a1dab To: Contact: Call-ID: 1cf7d0d159b488666bfe32db5fa28c3f@minixel CSeq: 102 OPTIONS User-Agent: Cisco 3845 Date: Thu, 01 May 2008 22:26:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP://66.28.147.204:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 66.28.190.219:5060;branch=z9hG4bK63d8320d;received=66.28.190.219;rport=5060 From: "Cisco 3845" ;tag=as3c1a1dab To: ;tag=as51c0b914 Call-ID: 1cf7d0d159b488666bfe32db5fa28c3f@minixel CSeq: 102 OPTIONS User-Agent: X-PRO release 1103g Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Accept: application/sdp Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '1cf7d0d159b488666bfe32db5fa28c3f@minixel' Method: OPTIONS Reliably Transmitting (no NAT) to 67.110.179.253:5060: OPTIONS sip:67.110.179.253 SIP/2.0 Via: SIP/2.0/UDP 66.28.190.219:5060;branch=z9hG4bK09ffa765;rport Max-Forwards: 70 From: "Cisco 3845" ;tag=as3edc0ee3 To: Contact: Call-ID: 307f90e567a6ed5857ae1a7d19189d18@minixel CSeq: 102 OPTIONS User-Agent: Cisco 3845 Date: Thu, 01 May 2008 22:26:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP://67.110.179.253:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 66.28.190.219:5060;branch=z9hG4bK09ffa765;rport From: "Cisco 3845" ;tag=as3edc0ee3 To: ;tag=4234798-13E3 Date: Thu, 01 May 2008 22:26:53 GMT Call-ID: 307f90e567a6ed5857ae1a7d19189d18@minixel Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 OPTIONS Supported: 100rel,resource-priority,replaces Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Accept: application/sdp Content-Type: application/sdp Content-Length: 457 v=0 o=CiscoSystemsSIP-GW-UserAgent 4865 7272 IN IP4 67.110.179.253 s=SIP Call c=IN IP4 67.110.179.253 t=0 0 m=audio 0 RTP/AVP 18 0 8 4 2 15 3 c=IN IP4 67.110.179.253 m=image 0 udptl t38 c=IN IP4 67.110.179.253 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy <-------------> --- (14 headers 18 lines) --- Really destroying SIP dialog '307f90e567a6ed5857ae1a7d19189d18@minixel' Method: OPTIONS RTP Debugging Enabled *CLI> Reliably Transmitting (NAT) to 72.187.243.97:45990: OPTIONS sip:7864335989@72.187.243.97:45990 SIP/2.0 Via: SIP/2.0/UDP 66.28.190.219:5060;branch=z9hG4bK2b81d934 Max-Forwards: 70 From: "Cisco 3845" ;tag=as73d8e490 To: Contact: Call-ID: 51506c26742cf04d7466b25854397863@minixel CSeq: 102 OPTIONS User-Agent: Cisco 3845 Date: Thu, 01 May 2008 22:26:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP://72.187.243.97:45990 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 66.28.190.219:5060;branch=z9hG4bK2b81d934 From: "Cisco 3845" To: ;tag=2816402910 Contact: Call-ID: 51506c26742cf04d7466b25854397863@minixel Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,REFER,NOTIFY CSeq: 102 OPTIONS Server: X-PRO release 1105x Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '51506c26742cf04d7466b25854397863@minixel' Method: OPTIONS <--- SIP read from UDP://72.187.243.97:45990 ---> <-------------> <--- SIP read from UDP://72.187.243.97:45990 ---> REGISTER sip:66.28.190.219 SIP/2.0 Via: SIP/2.0/UDP 72.187.243.97:45990;branch=z9hG4bKDFDC905DB9AE474CB70784E26CAE1C73 From: 7864335989 ;tag=1867072463 To: 7864335989 Contact: "7864335989" Call-ID: FF611B51FF2F4A3B9715B926FB8E7F7A@66.28.190.219 CSeq: 65632 REGISTER Expires: 60 Max-Forwards: 70 User-Agent: X-PRO release 1105x Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 72.187.243.97 : 45990 (NAT) <--- Transmitting (NAT) to 72.187.243.97:45990 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 72.187.243.97:45990;branch=z9hG4bKDFDC905DB9AE474CB70784E26CAE1C73;received=72.187.243.97 From: 7864335989 ;tag=1867072463 To: 7864335989 ;tag=as36f75e73 Call-ID: FF611B51FF2F4A3B9715B926FB8E7F7A@66.28.190.219 CSeq: 65632 REGISTER Server: Cisco 3845 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Expires: 60 Contact: ;expires=60 Date: Thu, 01 May 2008 22:26:56 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'FF611B51FF2F4A3B9715B926FB8E7F7A@66.28.190.219' in 32000 ms (Method: REGISTER) *CLI> *CLI> *CLI> *CLI> *CLI> <--- SIP read from UDP://67.203.64.22:5060 ---> OPTIONS sip:66.28.190.219:5060 SIP/2.0 Via: SIP/2.0/UDP 67.203.64.22:5060;branch=z9hG4bK48fkr5103oq00eoc4080 Call-ID: 9d5799d46b4aba74d5fe4b405585791e000cg60@192.168.4.50 To: sip:ping@66.28.190.219 From: ;tag=6ea35ee0741041b622ef91bd79a20d47000cg60 Max-Forwards: 70 CSeq: 1549 OPTIONS <-------------> --- (7 headers 0 lines) --- Looking for s in default (domain 66.28.190.219) <--- Transmitting (NAT) to 67.203.64.22:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 67.203.64.22:5060;branch=z9hG4bK48fkr5103oq00eoc4080;received=67.203.64.22 From: ;tag=6ea35ee0741041b622ef91bd79a20d47000cg60 To: sip:ping@66.28.190.219;tag=as64d699ca Call-ID: 9d5799d46b4aba74d5fe4b405585791e000cg60@192.168.4.50 CSeq: 1549 OPTIONS Server: Cisco 3845 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '9d5799d46b4aba74d5fe4b405585791e000cg60@192.168.4.50' in 32000 ms (Method: OPTIONS) <--- SIP read from UDP://67.203.64.22:5060 ---> OPTIONS sip:66.28.190.219:5060 SIP/2.0 Via: SIP/2.0/UDP 67.203.64.22:5060;branch=z9hG4bKp0ku31304gjh7e8sm641 Call-ID: 9d5799d46b4aba74d5fe4b405585791e000c070@192.168.4.50 To: sip:ping@66.28.190.219 From: ;tag=6ea35ee0741041b622ef91bd79a20d47000c070 Max-Forwards: 70 CSeq: 1550 OPTIONS <-------------> --- (7 headers 0 lines) --- Looking for s in default (domain 66.28.190.219) <--- Transmitting (NAT) to 67.203.64.22:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 67.203.64.22:5060;branch=z9hG4bKp0ku31304gjh7e8sm641;received=67.203.64.22 From: ;tag=6ea35ee0741041b622ef91bd79a20d47000c070 To: sip:ping@66.28.190.219;tag=as35ba9d84 Call-ID: 9d5799d46b4aba74d5fe4b405585791e000c070@192.168.4.50 CSeq: 1550 OPTIONS Server: Cisco 3845 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '9d5799d46b4aba74d5fe4b405585791e000c070@192.168.4.50' in 32000 ms (Method: OPTIONS) *CLI> *CLI> *CLI> Really destroying SIP dialog 'FF611B51FF2F4A3B9715B926FB8E7F7A@66.28.190.219' Method: REGISTER <--- SIP read from UDP://67.110.179.253:5060 ---> INVITE sip:7864335989@66.28.190.219:5060 SIP/2.0 Via: SIP/2.0/UDP 67.110.179.253:5060;branch=z9hG4bK7412924EF Remote-Party-ID: ;party=calling;screen=no;privacy=off From: ;tag=4231628-53 To: "7864335989" ;tag=as0da18e6c Date: Thu, 01 May 2008 22:27:29 GMT Call-ID: 3f9f6b31343bbd6b1ef7ab55068e23d5@minixel Supported: 100rel,timer,resource-priority,replaces Min-SE: 1800 Cisco-Guid: 2115709703-386142685-2860819775-2590138955 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Max-Forwards: 15 Timestamp: 1209680849 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 245 v=0 o=CiscoSystemsSIP-GW-UserAgent 7528 3553 IN IP4 67.110.179.253 s=SIP Call c=IN IP4 66.28.190.225 t=0 0 m=audio 10154 RTP/AVP 18 c=IN IP4 66.28.190.225 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 <-------------> --- (20 headers 11 lines) --- Sending to 67.110.179.253 : 5060 (NAT) <--- Transmitting (NAT) to 67.110.179.253:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 67.110.179.253:5060;branch=z9hG4bK7412924EF;received=67.110.179.253 From: ;tag=4231628-53 To: "7864335989" ;tag=as0da18e6c Call-ID: 3f9f6b31343bbd6b1ef7ab55068e23d5@minixel CSeq: 101 INVITE Server: Cisco 3845 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> Audio is at 66.28.190.219 port 28352 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 67.110.179.253:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 67.110.179.253:5060;branch=z9hG4bK7412924EF;received=67.110.179.253 From: ;tag=4231628-53 To: "7864335989" ;tag=as0da18e6c Call-ID: 3f9f6b31343bbd6b1ef7ab55068e23d5@minixel CSeq: 101 INVITE Server: Cisco 3845 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 291 v=0 o=root 225492555 225492555 IN IP4 66.28.190.219 s=Cisco-SIPGateway/IOS-12.x c=IN IP4 66.28.190.219 t=0 0 m=audio 28352 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> <--- SIP read from UDP://67.110.179.253:5060 ---> ACK sip:7864335989@66.28.190.219:5060 SIP/2.0 Via: SIP/2.0/UDP 67.110.179.253:5060;branch=z9hG4bK7412A446 From: ;tag=4231628-53 To: "7864335989" ;tag=as0da18e6c Date: Thu, 01 May 2008 22:27:29 GMT Call-ID: 3f9f6b31343bbd6b1ef7ab55068e23d5@minixel Max-Forwards: 15 CSeq: 101 ACK Allow-Events: telephone-event Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from UDP://67.110.179.253:5060 ---> BYE sip:7864335989@66.28.190.219:5060 SIP/2.0 Via: SIP/2.0/UDP 67.110.179.253:5060;branch=z9hG4bK7412B1CD3 From: ;tag=4231628-53 To: "7864335989" ;tag=as0da18e6c Date: Thu, 01 May 2008 22:27:29 GMT Call-ID: 3f9f6b31343bbd6b1ef7ab55068e23d5@minixel User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 15 Timestamp: 1209680849 CSeq: 102 BYE Max-Forwards: 70 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 67.110.179.253 : 5060 (NAT) <--- Transmitting (NAT) to 67.110.179.253:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 67.110.179.253:5060;branch=z9hG4bK7412B1CD3;received=67.110.179.253 From: ;tag=4231628-53 To: "7864335989" ;tag=as0da18e6c Call-ID: 3f9f6b31343bbd6b1ef7ab55068e23d5@minixel CSeq: 102 BYE Server: Cisco 3845 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> *CLI> *CLI> *CLI> stop now Really destroying SIP dialog '9d5799d46b4aba74d5fe4b405585791e000cg60@192.168.4.50' Method: OPTIONS Really destroying SIP dialog '9d5799d46b4aba74d5fe4b405585791e000c070@192.168.4.50' Method: OPTIONS <--- SIP read from UDP://72.187.243.97:45990 ---> BYE sip:14166289921@66.28.190.219 SIP/2.0 Via: SIP/2.0/UDP 72.187.243.97:45990;branch=z9hG4bK8F04E852858642CB8BF95CACDA1063A8 From: 7864335989 ;tag=1888197430 To: ;tag=as5822860e Contact: Call-ID: 99A84C56-D09C-498F-B98D-36F1B01D591A@192.168.1.111 CSeq: 5599 BYE Max-Forwards: 70 User-Agent: X-PRO release 1105x Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [May 1 18:27:45] ERROR[7307]: chan_sip.c:19213 handle_request_do: We could NOT get the channel lock for SIP/7864335989-11c76218! [May 1 18:27:45] ERROR[7307]: chan_sip.c:19214 handle_request_do: SIP transaction failed: 99A84C56-D09C-498F-B98D-36F1B01D591A@192.168.1.111 <--- Transmitting (NAT) to 72.187.243.97:45990 ---> SIP/2.0 503 Server error Via: SIP/2.0/UDP 72.187.243.97:45990;branch=z9hG4bK8F04E852858642CB8BF95CACDA1063A8;received=72.187.243.97 From: 7864335989 ;tag=1888197430 To: ;tag=as5822860e Call-ID: 99A84C56-D09C-498F-B98D-36F1B01D591A@192.168.1.111 CSeq: 5599 BYE Server: Cisco 3845 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> Reliably Transmitting (no NAT) to 66.28.147.204:5060: OPTIONS sip:66.28.147.204 SIP/2.0 Via: SIP/2.0/UDP 66.28.190.219:5060;branch=z9hG4bK7cefd8a4;rport Max-Forwards: 70 From: "Cisco 3845" ;tag=as4cec3036 To: Contact: Call-ID: 4490872d517e4e547219d32e3b3d90f3@minixel CSeq: 102 OPTIONS User-Agent: Cisco 3845 Date: Thu, 01 May 2008 22:27:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP://66.28.147.204:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 66.28.190.219:5060;branch=z9hG4bK7cefd8a4;received=66.28.190.219;rport=5060 From: "Cisco 3845" ;tag=as4cec3036 To: ;tag=as7918ee59 Call-ID: 4490872d517e4e547219d32e3b3d90f3@minixel CSeq: 102 OPTIONS User-Agent: X-PRO release 1103g Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Accept: application/sdp Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '4490872d517e4e547219d32e3b3d90f3@minixel' Method: OPTIONS Reliably Transmitting (no NAT) to 67.110.179.253:5060: OPTIONS sip:67.110.179.253 SIP/2.0 Via: SIP/2.0/UDP 66.28.190.219:5060;branch=z9hG4bK7a6f83e6;rport Max-Forwards: 70 From: "Cisco 3845" ;tag=as72fa879e To: Contact: Call-ID: 0d91626037890e027aa0bc735bd28a35@minixel CSeq: 102 OPTIONS User-Agent: Cisco 3845 Date: Thu, 01 May 2008 22:27:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP://67.110.179.253:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 66.28.190.219:5060;branch=z9hG4bK7a6f83e6;rport From: "Cisco 3845" ;tag=as72fa879e To: ;tag=4243204-744 Date: Thu, 01 May 2008 22:27:53 GMT Call-ID: 0d91626037890e027aa0bc735bd28a35@minixel Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 OPTIONS Supported: 100rel,resource-priority,replaces Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Accept: application/sdp Content-Type: application/sdp Content-Length: 457 v=0 o=CiscoSystemsSIP-GW-UserAgent 3498 6857 IN IP4 67.110.179.253 s=SIP Call c=IN IP4 67.110.179.253 t=0 0 m=audio 0 RTP/AVP 18 0 8 4 2 15 3 c=IN IP4 67.110.179.253 m=image 0 udptl t38 c=IN IP4 67.110.179.253 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy <-------------> --- (14 headers 18 lines) --- Really destroying SIP dialog '0d91626037890e027aa0bc735bd28a35@minixel' Method: OPTIONS Reliably Transmitting (NAT) to 72.187.243.97:45990: OPTIONS sip:7864335989@72.187.243.97:45990 SIP/2.0 Via: SIP/2.0/UDP 66.28.190.219:5060;branch=z9hG4bK0558b47d Max-Forwards: 70 From: "Cisco 3845" ;tag=as2d04d45b To: Contact: Call-ID: 5ff2bdc418349b510b1f41052b4cded5@minixel CSeq: 102 OPTIONS User-Agent: Cisco 3845 Date: Thu, 01 May 2008 22:27:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP://72.187.243.97:45990 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 66.28.190.219:5060;branch=z9hG4bK0558b47d From: "Cisco 3845" To: ;tag=2888884057 Contact: Call-ID: 5ff2bdc418349b510b1f41052b4cded5@minixel Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,REFER,NOTIFY CSeq: 102 OPTIONS Server: X-PRO release 1105x Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '5ff2bdc418349b510b1f41052b4cded5@minixel' Method: OPTIONS <--- SIP read from UDP://72.187.243.97:45990 ---> REGISTER sip:66.28.190.219 SIP/2.0 Via: SIP/2.0/UDP 72.187.243.97:45990;branch=z9hG4bKF3A4A8C5CE3646489ED8486A2FC43AB8 From: 7864335989 ;tag=1867072463 To: 7864335989 Contact: "7864335989" Call-ID: FF611B51FF2F4A3B9715B926FB8E7F7A@66.28.190.219 CSeq: 65633 REGISTER Expires: 60 Max-Forwards: 70 User-Agent: X-PRO release 1105x Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 72.187.243.97 : 45990 (NAT) <--- Transmitting (NAT) to 72.187.243.97:45990 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 72.187.243.97:45990;branch=z9hG4bKF3A4A8C5CE3646489ED8486A2FC43AB8;received=72.187.243.97 From: 7864335989 ;tag=1867072463 To: 7864335989 ;tag=as371087c1 Call-ID: FF611B51FF2F4A3B9715B926FB8E7F7A@66.28.190.219 CSeq: 65633 REGISTER Server: Cisco 3845 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Expires: 60 Contact: ;expires=60 Date: Thu, 01 May 2008 22:27:56 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'FF611B51FF2F4A3B9715B926FB8E7F7A@66.28.190.219' in 32000 ms (Method: REGISTER) <--- SIP read from UDP://67.203.64.22:5060 ---> OPTIONS sip:66.28.190.219:5060 SIP/2.0 Via: SIP/2.0/UDP 67.203.64.22:5060;branch=z9hG4bKcfg9531050d1ffsee141 Call-ID: 9d5799d46b4aba74d5fe4b405585791e000cg70@192.168.4.50 To: sip:ping@66.28.190.219 From: ;tag=6ea35ee0741041b622ef91bd79a20d47000cg70 Max-Forwards: 70 CSeq: 1551 OPTIONS <-------------> --- (7 headers 0 lines) --- Looking for s in default (domain 66.28.190.219) <--- Transmitting (NAT) to 67.203.64.22:5060 ---> SIP/2.0 503 Unavailable Via: SIP/2.0/UDP 67.203.64.22:5060;branch=z9hG4bKcfg9531050d1ffsee141;received=67.203.64.22 From: ;tag=6ea35ee0741041b622ef91bd79a20d47000cg70 To: sip:ping@66.28.190.219;tag=as7c4b8794 Call-ID: 9d5799d46b4aba74d5fe4b405585791e000cg70@192.168.4.50 CSeq: 1551 OPTIONS Server: Cisco 3845 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '9d5799d46b4aba74d5fe4b405585791e000cg70@192.168.4.50' in 32000 ms (Method: OPTIONS)